Audio signal coding method, decoding method, audio signal coding apparatus, and decoding apparatus where first vector quantization is performed on a signal and second vector quantization is performed on an error component resulting from the first vector quantization

ABSTRACT

A coding unit codes an audio signal by using a vector quantization method to reduce the quantity of data. An audio code having a minimum distance among auditive distances between sub-vectors produced by dividing an input vector and audio codes in a transmission-side code book is selected. A portion corresponding to an element of a sub-vector having a high auditive importance is handled in an audio code selecting unit while neglecting the codes indicating phase information and subjected to comparative retrieval with respect to audio codes in a transmission-side code book. Extracted phase information corresponding to an element portion of the sub-vector is added to the result obtained and output as a code index. Thereby, the calculation amount in the code retrieval of vector quantization and the number of codes in the code book are decreased without lowering the quality of an audio signal.

TECHNICAL FIELD

The present invention relates to coding apparatuses and methods in whicha feature quantity obtained from an audio signal such as a voice signalor a music signal, especially a signal obtained by transforming an audiosignal from time-domain to frequency-domain using a method likeorthogonal transformation, is efficiently coded so that it is expressedwith fewer coded streams as compared with the original audio signal, andto decoding apparatuses and methods having a structure capable ofdecoding a high-quality and broad-band audio signal using all or only aportion of the coded streams which are coded signals.

Various methods for efficiently coding and decoding audio signals havebeen proposed. Especially for an audio signal having a frequency bandexceeding 20 kHz such as a music signal, an MPEG audio method has beenproposed in recent years. In the coding method represented by the MPEGmethod, a digital audio signal on the time axis is transformed to dataon the frequency axis using orthogonal transform such as cosinetransform, and data on the frequency axis are coded from auditivelyimportant data by using the auditive sensitivity characteristic of humanbeings, whereas auditively unimportant data and redundant data are notcoded. In order to express an audio signal with a data quantityconsiderably smaller than the data quantity of the original digitalsignal, there is a coding method using a vector quantization method,such as TC-WVQ. The MPEG audio and the TC-WVQ are described in “ISO/IECstandard IS-11172-3” and “T. Moriya, H. Suga: An 8 Kbits transform coderfor noisy channels, Proc. ICASSP 89, pp. 196-199”, respectively.Hereinafter, the structure of a conventional audio coding apparatus willbe explained using FIG. 37. In FIG. 37, reference numeral 1601 denotesan FFT unit which frequency-transforms an input signal, 1602 denotes anadaptive bit allocation calculating unit which codes a specific band ofthe frequency-transformed input signal, 1603 denotes a sub-band divisionunit which divides the input signal into plural bands, 1604 denotes ascale factor normalization unit which normalizes the plural bandcomponents, and 1605 denotes a scalar quantization unit.

A description is given of the operation. An input signal is input to theFFT unit 1601 and the sub-band division unit 1603. In the FFT unit 1601,the input signal is subjected to frequency transformation, and input tothe adaptive bit allocation unit 1602; In the adaptive bit allocationunit 1602, how much data quantity is to be given to a specific bandcomponent is calculated on the basis of the minimum audible limit, whichis defined according to the auditive characteristic of human beings andthe masking characteristic, and the data quantity allocation for eachband is coded as an index.

On the other hand, in the sub-band division unit 1603, the input signalis divided into, for example, 32 bands, to be output. In the scalefactor normalization unit 1604, for each band component obtained in thesub-band division unit 1603, normalization is carried out with arepresentative value. The normalized value is quantized as an index. Inthe scalar quantization unit 1605, on the basis of the bit allocationcalculated by the adaptive bit allocation calculating unit 1602, theoutput from the scale factor normalization unit 1604 isscalar-quantized, and the quantized value is coded as an index.

Meanwhile, various methods of efficiently coding an acoustic signal havebeen proposed. Especially in recent years, a signal having a frequencyband of about 20 kHz, such as a music signal, is coded using the MPEGaudio method or the like. In the methods represented by the MPEG method,a digital audio signal on the time axis is transformed to the frequencyaxis using orthogonal transform, and data on the frequency axis aregiven data quantities, with a priority to auditively important one,while considering the auditive sensitivity characteristic of humanbeings. In order to express a signal having a data quantity considerablysmaller than the data quantity of the original digital signal, employedis a coding method using a vector quantization method, such as TCWVQ(Transform Coding for Weighted Vector Quantization). The MPEG audio andthe TCWVQ are described in “ISO/IEC standard IS-11172-3” and “T. Moriya,H. Suga: An 8 Kbits transform coder for noisy channels, Proc. ICASSP 89,pp. 196-199”, respectively.

In the conventional audio signal coding apparatus constructed asdescribed above, it is general that the MPEG audio method is used sothat coding is carried out with a data quantity of 64000 bits/sec foreach channel. With a data quantity smaller than this, the reproduciblefrequency band width and the subjective quality of decoded audio signalare sometimes degraded considerably. The reason is as follows. As in theexample shown in FIG. 37, the coded data are roughly divided into threemain parts, i.e., the bit allocation, the band representative value, andthe quantized value. So, when the compression ratio is high, asufficient data quantity is not allocated to the quantized value.Further, in the conventional audio signal coding apparatus, it isgeneral that a coder and a decoder are constructed with the dataquantity to be coded and the data quantity to be decoded being equal toeach other. For example, in a method where a data quantity of 128000bits/sec is coded, a data quantity of 128000 bits is decoded in thedecoder.

However, in the conventional audio signal coding and decodingapparatuses, coding and decoding must be carried out with a fixed dataquantity to obtain a good sound quality and, therefore, it is impossibleto obtain a high-quality sound at a high compression ratio.

The present invention is made to solve the above-mentioned problems andhas for its object to provide audio signal coding and decodingapparatuses, and audio signal coding and decoding methods, in which ahigh quality and a broad reproduction frequency band are obtained evenwhen coding and decoding are carried out with a small data quantity and,further, the data quantity in the coding and decoding can be variable,not fixed.

Furthermore, in the conventional audio signal coding apparatus,quantization is carried out by outputting a code index corresponding toa code that provides a minimum auditive distance between each codepossessed by a code block and an audio feature vector. However, when thenumber of codes possessed by the code book is large, the calculationamount significantly increases when retrieving an optimum code. Further,when the data quantity possessed by the code book is large, a largequantity of memory is required when the coding apparatus is constructedby hardware, and this is uneconomical. Further, on the receiving end,retrieval and memory quantity corresponding to the code indices arerequired.

The present invention is made to solve the above-mentioned problems andhas for its object to provide an audio signal coding apparatus thatreduces the number of times of code retrieval, and efficiently quantizesan audio signal with a code book having a lower number of codes, and anaudio signal decoding apparatus that can decode the audio signal.

DISCLOSURE OF THE INVENTION

An audio signal coding method according to the present invention is amethod for coding a data quantity by vector quantization using amultiple-stage quantization method comprising a first-stage vectorquantization process for vector-quantizing a frequency characteristicsignal sequence which is obtained by frequency transformation of aninput audio signal, and second-and-onward-stages of vector quantizationprocesses for vector-quantizing a quantization error component in theprevious-stage vector quantization process: wherein, among the multiplestages of quantization processes according to the multiple-stagequantization method, at least one vector quantization process performsvector quantization using, as weighting coefficients for quantization,weighting coefficients on frequency, calculated on the basis of thespectrum of the input audio signal and the auditive sensitivitycharacteristic showing the auditive nature of human beings.

Another audio signal method according to the present invention is amethod for coding a data quantity by vector quantization using amultiple-stage quantization method comprising a first vectorquantization process for vector-quantizing a frequency characteristicsignal sequence which is obtained by frequency transformation of aninput audio signal, and a second vector quantization process forvector-quantizing a quantization error component in the first vectorquantization process: wherein, on the basis of the spectrum of the inputaudio signal and the auditive sensitivity characteristic showing theauditive nature of human beings, a frequency block having a highimportance for quantization is selected from frequency blocks of thequantization error component in the first vector quantization processand, in the second vector quantization process, the quantization errorcomponent of the first quantization process is quantized with respect tothe selected frequency block.

Another audio signal coding method according to the present invention isa method for coding a data quantity by vector quantization using amultiple-stage quantization method comprising a first-stage vectorquantization process for vector-quantizing a frequency characteristicsignal sequence which is obtained by frequency transformation of aninput audio signal, and second-and-onward-stages of vector quantizationprocesses for vector-quantizing a quantization error component in theprevious-stage vector quantization process: wherein, among the multiplestages of quantization processes according to the multiple-stagequantization method, at least one vector quantization process performsvector quantization using, as weighting coefficients for quantization,weighting coefficients on frequency, calculated on the basis of thespectrum of the input audio signal and the auditive sensitivitycharacteristic showing the auditive nature of human beings; and, on thebasis of the spectrum of the input audio signal and the auditivesensitivity characteristic showing the auditive nature of human beings,a frequency block having a high importance for quantization is selectedfrom frequency blocks of the quantization error component in thefirst-stage vector quantization process and, in the second-stage vectorquantization process, the quantization error component of thefirst-stage quantization process is quantized with respect to theselected frequency block.

Another audio signal coding apparatus according to the present inventioncomprises: a time-to-frequency transformation unit for transforming aninput audio signal to a frequency-domain signal, a spectrum envelopecalculation unit for calculating a spectrum envelope of the input audiosignal; a normalization unit for normalizing the frequency-domain signalobtained in the time-to-frequency transformation unit, with the spectrumenvelope obtained in the spectrum envelope calculation unit, thereby toobtain a residual signal; an auditive weighting calculation unit forcalculating weighting coefficients on frequency, on the basis of thespectrum of the input audio signal and the auditive sensitivitycharacteristic showing the auditive nature of human beings; and amultiple-stage quantization unit having multiple stages of vectorquantization units connected in columns, to which the normalizedresidual signal is input, at least one of the vector quantization unitsperforming quantization using weighting coefficients obtained in theweighting unit.

Another audio signal coding apparatus according to the present inventionincludes plural quantization units among the multiple stages of themultiple-stage quantization unit that perform quantization using theweighting coefficients obtained in the weighting unit, and the auditiveweighting calculation unit calculates individual weighting coefficientsto be used by the multiple stages of quantization units, respectively.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein the multiple-stagequantization unit comprises: a first-stage quantization unit forquantizing the residual signal normalized by the normalization unit,using the spectrum envelope obtained in the spectrum envelopecalculation unit as weighting coefficients in the respective frequencydomains; a second-stage quantization unit for quantizing a quantizationerror signal from the first-stage quantization unit, using weightingcoefficients calculated on the basis of the correlation between thespectrum envelope and the quantization error signal of the first-stagequantization unit, as weighting coefficients in the respective frequencydomains, and a third-stage quantization unit for quantizing aquantization error signal from the second-stage quantization unit using,as weighting coefficients in the respective frequency domains, weightingcoefficients which are obtained by adjusting the weighting coefficientscalculated by the auditive weighting calculating unit according to theinput signal transformed to the frequency-domain signal by thetime-to-frequency transformation unit and the auditive characteristic,on the basis of the spectrum envelope, the quantization error signal ofthe second-stage quantization unit, and the residual signal normalizedby the normalization unit.

Another audio signal coding apparatus according to the present inventioncomprises: a time-to-frequency transformation unit for transforming aninput audio signal to a frequency-domain signal, a spectrum envelopecalculation unit for calculating a spectrum envelope of the input audiosignal; a normalization unit for normalizing the frequency-domain signalobtained in the time-to-frequency transformation unit, with the spectrumenvelope obtained in the spectrum envelope calculation unit, thereby toobtain a residual signal; a first vector quantizer for quantizing theresidual signal normalized by the normalization unit; an auditiveselection means for selecting a frequency block having a high importancefor quantization among frequency blocks of the quantization errorcomponent of the first vector quantizer, on the basis of the spectrum ofthe input audio signal and the auditive sensitivity characteristicshowing the auditive nature of human beings; and a second quantizer forquantizing the quantization error component of the first vectorquantizer with respect to the frequency block selected by the auditiveselection means.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein the auditive selection meansselects a frequency block using, as a scale of importance to bequantized, a value obtained by multiplying the quantization errorcomponent of the first vector quantizer, the spectrum envelope signalobtained in the spectrum envelope calculation unit, and an inversecharacteristic of the minimum audible limit characteristic.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein the auditive selection meansselects a frequency block using, as a scale of importance to bequantized, a value obtained by multiplying the spectrum envelope signalobtained in the spectrum envelope calculation unit and an inversecharacteristic of the minimum audible limit characteristic.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein the auditive selection meansselects a frequency block using, as a scale of importance to bequantized, a value obtained by multiplying the quantization errorcomponent of the first vector quantizer, the spectrum envelope signalobtained in the spectrum envelope calculation unit, and an inversecharacteristic of a characteristic obtained by adding the minimumaudible limit characteristic and a masking characteristic calculatedfrom the input signal.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein the auditive selection meansselects a frequency block using, as a scale of importance to bequantized, a value obtained by multiplying the quantization errorcomponent of the first vector quantizer, the spectrum envelope signalobtained in the spectrum envelope calculation unit, and an inversecharacteristic of a characteristic obtained by adding the minimumaudible limit characteristic and a masking characteristic that iscalculated from the input signal and corrected according to the residualsignal normalized by the normalization unit, the spectrum envelopesignal obtained in the spectrum envelope calculation unit, and thequantization error signal of the first-stage quantization unit.

An audio signal coding apparatus according to the present invention isan apparatus for coding a data quantity by vector quantization using amultiple-stage quantization means comprising a first vector quantizerfor vector-quantizing a frequency characteristic signal sequenceobtained by frequency transformation of an input audio signal, and asecond vector quantizer for vector-quantizing a quantization errorcomponent of the first vector quantizer: wherein the multiple-stagequantization means divides the frequency characteristic signal sequenceinto coefficient streams corresponding to at least two frequency bands,and each of the vector quantizers performs quantization, independently,using a plurality of divided vector quantizers which are preparedcorresponding to the respective coefficient streams.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus further comprising a normalizationmeans for normalizing the frequency characteristic signal sequence.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein the quantization meansappropriately selects a frequency band having a largeenergy-addition-sum of the quantization error, from the frequency bandsof the frequency characteristic signal sequence to be quantized, andthen quantizes the selected band.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein the quantization meansappropriately selects a frequency band from the frequency bands of thefrequency characteristic signal sequence to be quantized, on the basisof the auditive sensitivity characteristic showing the auditive natureof human beings, which frequency band selected has a largeenergy-addition-sum of the quantization error weighted by giving a largevalue to a band having a high importance of the auditive sensitivitycharacteristic, and then the quantization means quantizes the selectedband.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein the quantization means has avector quantizer serving as an entire band quantization unit whichquantizes, once at least, all of the frequency bands of the frequencycharacteristic signal sequence to be quantized.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein the quantization means isconstructed so that the first-stage vector quantizer calculates aquantization error in vector quantization using a vector quantizationmethod with a code book and, further, the second-stage quantizervector-quantizes the calculated quantization error.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein, as the vector quantizationmethod, code vectors, all or a portion of which codes are inverted, areused for code retrieval.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus further comprising a normalizationmeans for normalizing the frequency characteristic signal sequence,wherein calculation of distances used for retrieval of an optimum codein vector quantization is performed by calculating distances using, asweights, normalized components of the input signal processed by thenormalization unit, and extracting a code having a minimum distance.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein the distances are calculatedusing, as weights, both of the normalized components of the frequencycharacteristic signal sequence processed by the normalization means anda value in view of the auditive sensitivity characteristic showing theauditive nature of human beings, and a code having a minimum distance isextracted.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein the normalization means hasa frequency outline normalization unit that roughly normalizes theoutline of the frequency characteristic signal sequence.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein the normalization means hasa band amplitude normalization unit that divides the frequencycharacteristic signal sequence into a plurality of components ofcontinuous unit bands, and normalizes the signal sequence by dividingeach unit band with a single value.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein the quantization meansincludes a vector quantizer for quantizing the respective coefficientstreams of the frequency characteristic signal sequence independently bydivided vector quantizers, and includes a vector quantizer serving as anentire band quantization unit that quantizes, once at least, all of thefrequency bands of the input signal to be quantized.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein the quantization meanscomprises a first vector quantizer comprising a low-band divided vectorquantizer, an intermediate-band divided vector quantizer, and ahigh-band divided vector quantizer, and a second vector quantizerconnected after the first quantizer, and a third vector quantizerconnected after the second quantizer. The frequency characteristicsignal sequence input to the quantization means is divided into threebands, and the frequency characteristic signal sequence of low-bandcomponent among the three bands is quantized, by the low-band dividedvector quantizer. The frequency characteristic signal sequence ofintermediate-band component among the three bands is quantized by theintermediate-band divided vector quantizer, and the frequencycharacteristic signal sequence of high-band component among the threebands is quantized by the high-band divided vector quantizer,independently. A quantization error with respect to the frequencycharacteristic signal sequence is calculated in each of the dividedvector quantizers constituting the first vector quantizer, and thequantization error is input to the subsequent second vector quantizer.The second vector quantizer performs quantization for a band width to bequantized by the second vector quantizer, calculates a quantizationerror with respect to the input of the second vector quantizer, andinputs this to the third vector quantizer. The third vector quantizerperforms quantization for a band width to be quantized by the thirdvector quantizer.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus further comprising a firstquantization band selection unit between the first vector quantizer andthe second vector quantizer, and a second quantization band selectionunit between the second vector quantizer and the third vector quantizer:wherein the output from the first vector quantizer is input to the firstquantization band selection unit, and a band to be quantized by thesecond vector quantizer is selected in the first quantization bandselection unit. The second vector quantizer performs quantization for aband width to be quantized by the second vector quantizer, with respectto the quantization errors of the first three vector quantizers decidedby the first quantization band selection unit, calculates a quantizationerror with respect to the input to the second vector quantizer, andinputs this to the second quantization band selection unit. The secondquantization band selection unit selects a band to be quantized by thethird vector quantizer. The third vector quantizer performs quantizationfor a band decided by the second quantization band selection unit.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein, in place of the firstvector quantizer, the second vector quantizer or the third vectorquantizer is constructed using the low-band divided vector quantizer,the intermediate-band divided vector quantizer, and the high-banddivided vector quantizer.

Another audio signal decoding apparatus according to the presentinvention is an apparatus receiving, as an input, codes output from theaudio signal coding apparatus and decoding these codes to output asignal corresponding to the original input audio signal, and thisapparatus comprises: an inverse quantization unit for performing inversequantization using at least a portion of the codes output from thequantization means of the audio signal coding apparatus and an inversefrequency transformation unit for transforming a frequencycharacteristic signal sequence output from the inverse quantization unitto a signal corresponding to the original audio input signal.

Another audio signal decoding apparatus according to the presentinvention is an apparatus receiving, as an input, codes output from theaudio signal coding apparatus and decoding these codes to output asignal corresponding to the original input audio signal, and thisapparatus comprises: an inverse quantization unit for reproducing afrequency characteristic signal sequence; an inverse normalization unitfor reproducing normalized components on the basis of the codes outputfrom the audio signal coding apparatus, using the frequencycharacteristic signal sequence output from the inverse quantizationunit, and multiplying the frequency characteristic signal sequence andthe normalized components; and an inverse frequency transformation unitfor receiving the output from the inverse normalization unit andtransforming the frequency characteristic signal sequence to a signalcorresponding to the original audio signal.

Another audio signal decoding apparatus according to the presentinvention is an apparatus receiving, as an input, codes output from theaudio signal coding apparatus and decoding these codes to output asignal corresponding to the original audio signal, and this apparatuscomprises an inverse quantization unit which performs inversequantization using the output codes whether the codes are output fromall of the vector quantizers constituting the quantization means in theaudio signal coding apparatus or from some of them.

Another audio signal decoding apparatus according to the presentinvention is an audio signal decoding apparatus wherein the inversequantization unit performs inverse quantization of quantized codes in aprescribed band by executing, alternately, inverse quantization ofquantized codes in a next stage, and inverse quantization of quantizedcodes in a band different from the prescribed band. When there are noquantized codes in the next stage during the inverse quantization, theinverse quantization unit continuously executes the inverse quantizationof quantized codes in the different band and, when there are noquantized codes in the different band, the inverse quantization unitcontinuously executes the inverse quantization of quantized codes in thenext stage.

Another audio signal decoding apparatus according to the presentinvention is an apparatus receiving, as an input, codes output from theaudio signal coding apparatus and decoding these codes to output asignal corresponding to the original input audio signal, and thisapparatus comprises an inverse quantization unit which performs inversequantization using only codes output from the low-band divided vectorquantizer as a constituent of the first vector quantizer even though allor some of the three divided vector quantizers constituting the firstvector quantizer in the audio signal coding apparatus output codes.

Another audio signal decoding apparatus according to the present invention is an audio signal decoding apparatus wherein the inversequantization unit performs inverse quantization using codes output fromthe second vector quantizer, in addition to the codes output from thelow-band divided vector quantizer as a constituent of the first vectorquantizer.

Another audio signal decoding apparatus according to the presentinvention is an audio signal decoding apparatus wherein the inversequantization unit performs inverse quantization using codes output fromthe intermediate-band divided vector quantizer as a constituent of thefirst vector quantizer, in addition to the codes output from thelow-band divided vector quantizer as a constituent of the first vectorquantizer and the codes output from the second vector quantizer.

Another audio signal decoding apparatus according to the presentinvention is an audio signal decoding apparatus wherein the inversequantization unit performs inverse quantization using codes output fromthe third vector quantizer, in addition to the codes output from thelow-band divided vector quantizer as a constituent of the first vectorquantizer, the codes output from the second vector quantizer, and thecodes output from the intermediate-band divided vector quantizer as aconstituent of the first vector quantizer.

Another audio signal decoding apparatus according to the presentinvention is an audio signal decoding apparatus wherein the inversequantization unit performs inverse quantization using codes output fromthe high-band divided vector quantizer as a constituent of the firstvector quantizer, in addition to the codes output from the low-banddivided vector quantizer as a constituent of the first vector quantizer,the codes output from the second vector quantizer, the codes output fromthe intermediate-band divided vector quantizer as a constituent of thefirst vector quantizer, and the codes output from the third vectorquantizer.

Another audio signal coding apparatus according to the present inventioncomprises: a phase information extraction unit for receiving, as aninput signal, a frequency characteristic signal sequence by obtained byfrequency transformation of an input audio signal, and extracting phaseinformation of a portion of the frequency characteristic signal sequencecorresponding to a prescribed frequency band, a code book for containinga plurality of audio codes being representative values of the frequencycharacteristic signal sequence, wherein an element portion of each audiocode corresponding to the extracted phase information is shown by anabsolute value; and an audio code selection unit for calculating theauditive distances between the frequency characteristic signal sequenceand the respective audio codes in the code book, selecting an audio codehaving a minimum distance, adding phase information to the audio codehaving the minimum distance using the output from the phase informationextraction unit as auxiliary information, and outputting a code indexcorresponding to the audio code having the minimum distance as an outputsignal.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein the phase informationextraction unit extracts phase information of a prescribed number ofelements on the low-frequency band side of the input frequencycharacteristic signal sequence.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus further comprising an auditivepsychological weight vector table which is a table of auditivepsychological quantities relative to the respective frequencies in viewof the auditive psychological characteristic of human beings: whereinthe phase information extraction unit extracts phase information of anelement which matches with a vector stored in the auditive psychologicalweight vector table, from the input frequency characteristic signalsequence.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus further comprising a smoothing unitfor smoothing the frequency characteristic signal sequence using asmoothing vector by division between vector elements: wherein, beforeselecting the audio code having the minimum distance and adding thephase information to the selected audio code, the audio code selectingunit converts the selected audio code to an audio code which has notbeen subjected to smoothing using smoothing information output from thesmoothing unit, and outputs a code index corresponding to the audio codeas an output signal.

An audio signal coding apparatus according to the present invention isan audio signal coding apparatus further comprising: an auditivepsychological weight vector table which is a table of auditivepsychological quantities relative to the respective frequencies, in viewof the auditive psychological characteristic of human beings; asmoothing unit for smoothing the frequency characteristic signalsequence using a smoothing vector by division between vector elements;and a sorting unit for selecting a plurality of values obtained bymultiplying the values of the auditive psychological weight vector tableand the values of the smoothing vector table, in order of auditiveimportance, and outputting these values toward the audio code selectionunit.

Another audio signal coding apparatus according to the present inventionis an audio signal-coding apparatus wherein employed as the frequencycharacteristic signal sequence is a vector of which elements arecoefficients obtained by subjecting the audio signal to frequencytransformation.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein employed as the frequencycharacteristic signal sequence is a vector of which elements arecoefficients obtained by subjecting the audio signal to frequencytransformation.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein employed as the frequencycharacteristic signal sequence is a vector of which elements arecoefficients obtained by subjecting the audio signal to frequencytransformation.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein employed as the frequencycharacteristic signal sequence is a vector of which elements arecoefficients obtained by subjecting the audio signal to MDCT (ModifiedDiscrete Cosine Transformation).

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein employed as the frequencycharacteristic signal sequence is a vector of which elements arecoefficients obtained by subjecting the audio signal to MDCT (ModifiedDiscrete Cosine Transformation).

Another audio signal coding, apparatus according to the presentinvention is an audio signal coding apparatus wherein employed as thefrequency characteristic signal sequence is a vector of which elementsare coefficients obtained by subjecting the audio signal to MDCT(Modified Discrete Cosine Transformation).

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein employed as the smoothingvector is a vector of which elements are relative frequency responses inthe respective frequencies, which are calculated from linear predictioncoefficients obtained by subjecting the audio signal to linearprediction.

Another audio signal coding apparatus according to the present inventionis an audio signal coding apparatus wherein employed as the smoothingvector is a vector of which elements are relative frequency responses inthe respective frequencies, which are calculated from linear predictioncoefficients obtained by subjecting the audio signal to linearprediction.

Another audio signal decoding apparatus according to the presentinvention comprises: a phase information extraction unit for receiving,as an input signal, one of code indices obtained by quantizing frequencycharacteristic signal sequences which are feature quantities of an audiosignal, and extracting phase information of elements of the input codeindex corresponding to a prescribed frequency band; a code book forcontaining a plurality of frequency characteristic signal sequencescorresponding to the code indices, wherein an element portioncorresponding to the extracted phase information is shown by an absolutevalue, and an audio code selection unit for calculating the auditivedistances between the input code index and the respective frequencycharacteristic signal sequences in the code book, selecting a frequencycharacteristic signal sequence having a minimum distance, adding phaseinformation to the frequency characteristic signal sequence having theminimum distance using the output from the phase information extractionunit as auxiliary information, and outputting the frequencycharacteristic signal sequence corresponding to the input code index asan output signal.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a diagram illustrating an overview of the structure of audiosignal coding and decoding apparatuses according to a first embodimentof the present invention.

FIG. 2 is a block diagram illustrating an example of a normalizationunit as a constituent of the above-described audio signal codingapparatus.

FIG. 3 is a block diagram illustrating an example of a frequency outlinenormalization unit as a constituent of the above-described audio signalcoding apparatus.

FIG. 4 is a diagram illustrating the detailed structure of aquantization unit in the coding apparatus.

FIG. 5 is a block diagram illustrating the structure of an audio signalcoding apparatus according to a second embodiment of the presentinvention.

FIG. 6 is a block diagram illustrating the structure of an audio signalcoding apparatus according to a third embodiment of the presentinvention.

FIG. 7 is a block diagram illustrating the detailed structures of aquantization unit and an auditive selection unit in each stage of theaudio signal coding apparatus shown in FIG. 6.

FIG. 8 is a diagram for explaining the quantizing operation of thevector quantizer.

FIG. 9 is a diagram showing error signal zi, spectrum envelope I1, andminimum audible limit characteristic hi.

FIG. 10 is a block diagram illustrating the detailed structures of otherexamples of each quantization unit and an auditive selection unitincluded in the audio signal coding apparatus shown in FIG. 6.

FIG. 11 is a block diagram illustrating the detailed structures of stillother examples of each quantization unit and an auditive selection unitincluded in the audio signal coding apparatus shown in FIG. 6.

FIG. 12 is a block diagram illustrating the detailed structures offurther examples of each quantization unit and an auditive selectionunit included in the audio signal coding apparatus shown in FIG. 6.

FIG. 13 is a diagram illustrating an example of selection of a frequencyblock having the highest importance (length W).

FIG. 14 is a block diagram illustrating the structure of an audio signalcoding apparatus according to a fourth embodiment of the presentinvention.

FIG. 15 is a block diagram illustrating the structure of an audio signalcoding apparatus according to a fifth embodiment of the presentinvention.

FIG. 16 is a block diagram illustrating the structure of an audio signalcoding apparatus according to a sixth embodiment of the presentinvention.

FIG. 17 is a block diagram illustrating the structure of an audio signalcoding apparatus according to a seventh embodiment of the presentinvention.

FIG. 18 is a block diagram illustrating the structure of an audio signalcoding apparatus according to an eighth embodiment of the presentinvention.

FIG. 19 is a diagram for explaining the detailed operation ofquantization in each quantization unit included in the coding apparatusaccording to any of the first to eighth embodiments.

FIG. 20 is a diagram for explaining an audio signal decoding apparatusaccording to a ninth embodiment of the present invention.

FIG. 21 is a diagram for explaining the audio signal decoding apparatusaccording to the ninth embodiment of the present invention.

FIG. 22 is a diagram for explaining the audio signal decoding apparatusaccording to the ninth embodiment of the present invention.

FIG. 23 is a diagram for explaining the audio signal decoding apparatusaccording to the ninth embodiment of the present invention.

FIG. 24 is a diagram for explaining the audio signal decoding apparatusaccording to the ninth embodiment of the present invention.

FIG. 25 is a diagram for explaining the audio signal decoding apparatusaccording to the ninth embodiment of the present invention.

FIG. 26 is a diagram for explaining the detailed operation of an inversequantization unit as a constituent of the audio signal decodingapparatus.

FIG. 27 is a diagram for explaining the detailed operation of an inversenormalization unit as a constituent of the audio signal decodingapparatus.

FIG. 28 is a diagram for explaining the detailed operation of afrequency outline inverse normalization unit as a constituent of theaudio signal decoding apparatus.

FIG. 29 is a diagram illustrating the structure of an audio signalcoding apparatus according to a tenth embodiment of the presentinvention.

FIG. 30 is a diagram for explaining the structure of an audio featurevector in the audio signal coding apparatus according to the tenthembodiment.

FIG. 31 is a diagram for explaining the processing of the audio signalcoding apparatus according to the tenth embodiment.

FIG. 32 is a diagram illustrating the detailed structure of an audiosignal coding apparatus according to an eleventh embodiment of thepresent invention, and an example of an auditive psychological weightvector table.

FIG. 33 is a diagram illustrating the detailed structure of an audiosignal coding apparatus according to a twelfth embodiment of the presentinvention, and for explaining the processing of a smoothing unit.

FIG. 34 is a diagram illustrating the detailed structure of an audiosignal coding apparatus according to a thirteenth embodiment of thepresent invention.

FIG. 35 is a diagram illustrating the detailed structure of an audiosignal coding apparatus according to a fourteenth embodiment of thepresent invention.

FIG. 36 is a diagram illustrating the structure of an audio signaldecoding apparatus according to a fifteenth embodiment of the presentinvention.

FIG. 37 is a diagram illustrating the structure of an audio signalcoding apparatus according to the prior art.

BEST MODES TO EXECUTE THE INVENTION Embodiment 1

FIG. 1 is a diagram illustrating an overview of the structure of audiosignal coding and decoding apparatuses according to a first embodimentof the invention. In FIG. 1, reference numeral 1 denotes a codingapparatus, and 2 denotes a decoding apparatus. In the coding apparatus1, reference numeral 101 denotes a frame division unit that divides aninput signal into a prescribed number of frames; 102 denotes a windowmultiplication unit that multiplies the input signal and a windowfunction on the time axis; 103 denotes an MDCT unit that performsmodified discrete cosine transform for time-to-frequency conversion of asignal on the time axis to a signal on the frequency axis; 104 denotes anormalization unit that receives both of the time axis signal outputfrom the frame division unit 101 and the MDCT coefficients output fromthe MDCT unit 103 and normalizes the MDCT coefficients; and 105 denotesa quantization unit that receives the normalized MDCT coefficients andquantizes them. Although MDCT is employed for time-to-frequencytransform in this embodiment, discrete Fourier transform (DFT) may beemployed.

In the decoding apparatus 2, reference numeral 106 denotes an inversequantization unit that receives a signal output from the codingapparatus 1 and inversely quantizes this signal; 107 denotes an inversenormalization unit that inversely normalizes the output from the inversequantization unit 106; 108 denotes an inverse MDCT unit that performsmodified discrete cosine transform of the output from the inversenormalization unit 107; 109 denotes a window multiplication unit; and110 denotes a frame overlapping unit.

A description is given of the operation of the audio signal coding anddecoding apparatuses constructed as described above.

It is assumed that the signal input to the coding apparatus 1 is adigital signal sequence that is temporally continuous. For example, itis a digital signal obtained by 16-bit quantization at a samplingfrequency of 48 kHz. This input signal is accumulated in the framedivision unit 101 until reaching a prescribed same number, and it isoutput when the accumulated sample number reaches a defined framelength. Here, the frame length of the frame division unit 101 is, forexample, any of 128, 256, 512, 1024, 2048, and 4096 samples. In theframe division unit 101, it is also possible to output the signal withthe frame length being variable according to the feature of the inputsignal. Further, the frame division unit 101 is constructed to performan output for each shift length specified. For example, in the casewhere the frame length is 4096 samples, when a shift length half as longas the frame length is set, the frame division unit 101 outputs latest4096 samples every time the frame length reaches 2048 samples. Ofcourse, even when the frame length or the sampling frequency varies, itis possible to have the structure in which the shift length is set athalf of the frame length.

The output from the frame division unit 101 is input to the windowmultiplication unit 102 and to the normalization unit 104. In the windowmultiplication unit 102, the output signal from the frame division unit101 is multiplied by a window function on the time axis, and the resultis output from the window multiplication unit 102. This manner is shownby, for example, formula (1).

hxi=hi·xi i=1, 2, . . . , N

$\begin{matrix}{{hi} = {\sin \quad \left( {\frac{\pi}{N}\quad \left( {i + 0.5} \right)} \right)}} & (1)\end{matrix}$

where xi is the output from the frame division unit 101, hi is thewindow function, and hxi is the output from the window multiplicationunit 102. Further, i is the suffix of time. The window function hi shownin formula (1) is an example, and the window function is not restrictedto that shown in formula (1). Selection of the window function dependson the feature of the input signal, the frame length of the framedivision unit 101, and the shapes of window functions in frames whichare located temporally before and after the frame being processed. Forexample, assuming that the frame length of the frame division unit 101is N, as the feature of the signal input to the window multiplicationunit 102, the average power of signals input at every N/4 is calculatedand, when the average power varies significantly, the calculation shownin formula (1) is executed with a frame length shorter than N. Further,it is desirable to appropriately select the window function, accordingto the shape of the window function of the previous frame and the shapeof the window function of the subsequent frame, so that the shape of thewindow function of the present frame is not distorted.

Next, the output from the window multiplication unit 102 is input to theMDCT unit 103, wherein modified discrete cosine transform is executed,and MDCT coefficients are output. A general formula of modified discretecosine transform is represented by formula (2).${yk} = {\sum\limits_{n = 0}^{N - 1}\quad {{{hx}_{n} \cdot \cos}\quad \left( \frac{2\quad \pi \quad \left( {k + {1/2}} \right)\quad \left( {n + n_{0}} \right)}{N} \right)}}$

 n ₀ =N/4+½ (k=0, 1, . . . , N/2−1)  (2)

Assuming that the MDCT coefficients output from the MDCT unit 103 areexpressed by yk in formula (2), the output from the MDCT unit 103 showsthe frequency characteristics, and it linearly corresponds to a lowerfrequency component as the variable k of yk approaches closer 0, whileit corresponds to a higher frequency component as the variable kapproaches closer N/2−1 from 0. The normalization unit 104 receives bothof the time axis signal output from the frame division unit 101 and theMDCT coefficients output from the MDCT unit 103, and normalizes the MDCTcoefficients using several parameters. To normalize the MDCTcoefficients is to suppress variations in values of the MDCTcoefficients, which values are considerably different between thelow-band component and the high-band component. For example, when thelow-band component is considerably larger than the high-band component,a parameter having a large value in the low-band component and a smallvalue in the high-band component is selected, and the MDCT coefficientsare divided by this parameter to suppress the variations of the MDCTcoefficients. In the normalization unit 104, the indices expressing theparameters used for the normalization are coded.

The quantization unit 105 receives the MDCT coefficients normalized bythe normalization unit 104, and quantizes the MDCT coefficients. Thequantization unit 105 codes indices expressing parameters used for thequantization.

On the other hand, in the decoding apparatus 2, decoding is carried outusing the indices from the normalization unit 104 in the codingapparatus 1, and the indices from the quantization unit 105. In theinverse quantization unit 106, the normalized MDCT coefficients arereproduced using the indices from the quantization unit 105. In theinverse quantization unit 106, the reproduction of the MDCT coefficientsmay be carried out using all or some of the indices. Of course, theoutput from the normalization unit 104 and the output from the inversequantization unit 106 are not always identical to those before thequantization because the quantization by the quantization unit 105 isattended with quantization errors.

In the inverse normalization unit 107, the parameters used for thenormalization in the coding apparatus 1 are restored from the indicesoutput from the normalization unit 104 of the coding apparatus 1, andthe output from the inverse quantization unit 106 is multiplied by thoseparameters to restore the MDCT coefficients. In the inverse MDCT unit108, the MDCT coefficients output from the inverse normalization unit107 are subjected to inverse MDCT, whereby the frequency-domain signalis restored to the time-domain signal. The inverse MDCT calculation isrepresented by, for example, formula (3).${{xx}\quad (n)} = {\frac{2}{N}\quad {\sum\limits_{k = 0}^{N - 1}\quad {{yy}_{k}\quad \cos \quad \left( \frac{2\quad \pi \quad \left( {k + {1/2}} \right)\quad \left( {n + n_{0}} \right)}{N} \right)}}}$

 n ₀ =N/4+½  (3)

where yyk is the MDCT coefficients restored in the inverse normalizationunit 107, and xx(k) is the inverse MDCT coefficients which are outputfrom the inverse MDCT unit 108.

The window multiplication unit 109 performs window multiplication usingthe output xx(k) from the inverse MDCT unit 108. The windowmultiplication is carried out using the same window as used by thewindow multiplication unit 102 of the coding apparatus B1, and a processshown by, for example, formula (4) is carried out.

z(i)=xx(i)·hi  (4)

where zi is the output from the window multiplication unit 109.

The frame overlapping unit 110 reproduces the audio signal using theoutput from the window multiplication unit 109. Since the output fromthe window multiplication unit 109 is a temporally overlapped signal,the frame overlapping unit 110 provides an output signal from thedecoding apparatus B2 using, for example, formula (5).

out(i)=z _(m)(i)+z _(m−1)(i+SHIFT)  (5)

where zm(i) is the i-th output signal z(i) from the windowmultiplication unit 109 in the m-th time frame, zm−1(i) is the i-thoutput signal from the window multiplication unit 19 in the (m−1)th timeframe, SHIFT is the sample number corresponding to the shift length ofthe coding apparatus, and out(i) is the output signal from the decodingapparatus 2 in the m-th time frame of the frame overlapping unit 110.

An example of the normalization unit 104 will be described in detailusing FIG. 2. In FIG. 2, reference numeral 201 denotes a frequencyoutline normalization unit that receives the outputs from the framedivision unit 101 and the MDCT unit 103; and 202 denotes a bandamplitude normalization unit that receives the output from the frequencyoutline normalization unit 201 and performs normalization with referenceto a band table 203.

A description is given of the operation. The frequency outlinenormalization unit 201 calculates a frequency outline, that is, a roughform of frequency, using the data on the time axis output from the framedivision unit 101, and divides the MDCT coefficients output from theMDCT unit 103 by this. Parameters used for expressing the frequencyoutline are coded as indices. The band amplitude normalization unit 202receives the output signal from the frequency outline normalization unit201, and performs normalization for each band shown in the band table203. For example, assuming that the MDCT coefficients output from thefrequency outline normalization unit 201 are dct(i) (i=0˜2047) and theband table 203 is, for example, as shown in Table 1, an average value ofamplitude in each band is calculated using, for example, formula (6).

TABLE 1 j bjlow bjhigh 0 0 10 1 11 22 2 23 33 3 34 45 4 46 56 5 57 68 669 80 7 81 92 8 93 104 9 105 116 10 117 128 11 129 141 12 142 153 13 154166 14 167 179 15 180 192 16 193 205 17 206 219 18 220 233 19 234 247 20248 261 21 262 276 22 277 291 23 292 307 24 308 323 25 324 339 26 340356 27 357 374 28 375 392 29 393 410 30 411 430 31 431 450 32 451 479 33471 492 34 493 515 35 516 538 36 539 563 37 564 588 38 589 615 39 616644 40 645 673 41 674 705 42 706 737 43 738 772 44 773 809 45 810 848 46849 889 47 890 932 48 933 978 49 979 1027 50 1028 1079 51 1080 1135 521136 1193 53 1194 1255 54 1256 1320 55 1321 1389 56 1390 1462 57 14631538 58 1539 1617 59 1618 1699 60 1700 1783 61 1784 1870 62 1871 1958 631959 2048

$\begin{matrix}{{\left. \begin{matrix}{{sum}_{j} = {\sum\limits_{i = {bjlow}}^{bjhigh}\quad {{dct}\quad (i)^{p}}}} \\{{ave}_{j} = \left( \frac{{sum}_{j}}{{bjhigh} - {bjlow} + 1} \right)^{- p}}\end{matrix} \right\} \quad {bjlow}} \leq i \leq {bjhigh}} & (6)\end{matrix}$

where bjlow and bjhigh are the lowest-band index i and the highest-bandindex i, respectively, in which dct(i) in the j-th band shown in theband table 203 belongs. Further, p is the norm in distance calculation,which is desired to be 2, and avej is the average of amplitude in eachband number j. The band amplitude normalization unit 202 quantizes theavej to obtain qavej, and normalizes it using, for example, formula (7).

n _(—) dct(i)=dct(i)/gave_(j) bjlow≦i≦bjhigh  (7)

To quantize the avej, scalar quantization may be employed, or vectorquantization may be carried out using the code book. The band amplitudenormalization unit 202 codes the indices of parameters used forexpressing the qavej.

Although the normalization unit 104 in the coding apparatus 1 isconstructed using both of the frequency outline normalization unit 201and the band amplitude normalization unit 202 as shown in FIG. 2, it maybe constructed using either of the frequency outline normalization unit201 and the band amplitude normalization unit 202. Further, when thereis no significant variation between the low-band component and thehigh-band component of the MDCT coefficients output from the MDCT unit103, the output from the MDCT unit 103 may be directly input to thequantization unit 105 without using the units 201 and 202.

The frequency outline normalization unit 201 shown in FIG. 2 will bedescribed in detail using FIG. 3. In FIG. 3, reference numeral 301denotes a linear predictive analysis unit that receives the output fromthe frame division unit 101 and performs linear predictive analysis; 302denotes an outline quantization unit that quantizes the coefficientobtained in the linear predictive analysis unit 301; and 303 denotes anenvelope characteristic normalization unit that normalizes the MDCTcoefficients by spectral envelope.

A description is given of the operation of the frequency outlinenormalization unit 201. The linear predictive analysis unit 301 receivesthe audio signal on the time axis from the frame division unit 101,performs linear predictive coding (LPC), and calculates linearpredictive coefficients (LPC coefficients). The linear predictivecoefficients can generally be obtained by calculating an autocorrelationfunction of a window-multiplied signal., such as Humming window, andsolving a normal equation or the like. The linear predictivecoefficients so calculated are converted to linear spectral paircoefficients (LSP coefficients) or the like and quantized in the outlinequantization unit 302. As a quantization method, vector quantization orscalar quantization may be employed. Then, frequency transfercharacteristic (spectral envelope) expressed by the parameters quantizedby the outline quantization unit 302 is calculated in the envelopecharacteristic normalization unit 303, and the MDCT coefficients outputfrom the MDCT unit 103 are divided by the characteristic to benormalized. To be specific, when the linear predictive coefficientsequivalent to the parameters quantized by the outline quantization unit302 are qlpc(i), the frequency transfer characteristic calculated by theenvelope characteristic normalization unit 303 is obtained by formula(8). $\begin{matrix}{{li} = \left\{ {{\begin{matrix}{{qlpc}\quad (i)} & {0 \leq i \leq {ORDER}} \\0 & {{{ORDER} + 1} \leq i < N}\end{matrix}{{env}(i)}} = {1/{{fft}({li})}}} \right.} & (8)\end{matrix}$

where ORDER is desired to be 10˜40, and fft( ) means high-speed Fouriertransform. Using the calculated frequency transfer characteristicenv(i), the envelope characteristic normalization unit 303 performsnormalization using, for example, formula (9) as follows.

fact(i)=mdct(i)/env(i)  (9)

where mdct(i) is the output signal from the MDCT unit 103, and fdct(i)is the normalized output signal from the envelope characteristicnormalization unit 303. Through the above-mentioned process steps, theprocess of normalizing the MDCT coefficient stream is completed.

Next, the quantization unit 105 in the coding apparatus 1 will bedescribed in detail using FIG. 4. In FIG. 4, reference numeral 4005denotes a multistage quantization unit that performs vector quantizationto the frequency characteristic signal sequence (MDCT coefficientstream) leveled by the normalization unit 104. The multistagequantization unit 4005 includes a first stage quantizer 40051, a secondstage quantizer 40052, . . . , an N-th stage quantizer 40053 which areconnected in a column. Further, 4006 denotes an auditive weightcalculating unit that receives the MDCT coefficients output from theMDCT unit 103 and the spectral envelope obtained in the envelopecharacteristic normalization unit 303, and provides a weightingcoefficient used for quantization in the multistage quantization unit4005, on the basis of the auditive sensitivity characteristic.

In the auditive weight calculating unit 4006, the MDCT coefficientstream output from the MDCT unit 103 and the LPC spectral envelopeobtained in the envelope characteristic normalization unit 303 are inputand, with respect to the spectrum of the frequency characteristic signalsequence output from the MDCT unit 103, on the basis of the auditivesensitivity characteristic which is the auditive nature of human beings,such as minimum audible limit characteristic and auditive maskingcharacteristic, a characteristic signal in regard to the auditivesensitivity characteristic is calculated and, furthermore, a weightingcoefficient used for quantization is obtained on the basis of thecharacteristic signal and the spectral envelope.

The normalized MDCT coefficients output from the normalization unit 104are quantized in the first stage quantizer 40051 in the multistagequantization unit 4005 using the weighting coefficient obtained by theauditive weight calculating unit 4006, and a quantization errorcomponent due to the quantization in the first stage quantizer 40051 isquantized in the second stage quantizer 40052 in the multistagequantization unit 4005 using the weighting coefficient obtained by theauditive weight calculating unit 4006. Thereafter, in the same manner asmentioned above, in each stage of the multistage quantization unit, aquantization error component due to quantization in the previous-stagequantizer is quantized. Coding of the audio signal is completed when aquantization error component due to quantization in the (N−1)th stagequantizer has been quantized in the N-th stage quantizer 40053 using theweighting coefficient obtained by the auditive weight calculating unit4006.

As described above, according to the audio signal coding apparatus ofthe first embodiment, vector quantization is carried out in the pluralstages of vector quantizers 40051˜40053 in the multistage quantizationmeans 4005 using, as a weight for quantization, a weighting coefficienton the frequency, which is calculated in the auditive weight calculatingunit 4006 on the basis of the spectrum of the input audio signal, theauditive sensitivity characteristic showing the auditive nature of humanbeings, and the LPC spectral envelope. Therefore, efficient quantizationcan be carried out utilizing the auditive nature of human beings.

In the audio signal coding apparatus shown in FIG. 4, the auditiveweight calculating unit 4006 uses the LPC spectral envelope forcalculation of the weighting coefficient. However, it may calculate theweighting coefficient using only the spectrum of input audio signal andthe auditive sensitivity characteristic showing the auditive nature ofhuman beings.

Further, in the audio signal coding apparatus shown in FIG. 4, all ofthe plural stages of vector quantizers in the multistage quantizationmeans 4005 perform quantization using the weighting coefficient obtainedin the auditive weight calculating unit 4006 on the basis of theauditive sensitivity characteristic. However, as long as any of theplural stages of vector quantizers in the multistage quantization means4005 performs quantization using the weighting coefficient on the basisof the auditive sensitivity characteristic, efficient quantization canbe carried out as compared with the case where such a weightingcoefficient on the basis of the auditive sensitivity characteristic isnot used.

Embodiment 2

FIG. 5 is a block diagram illustrating the structure of an audio signalcoding apparatus according to a second embodiment of the invention. Inthis embodiment, only the structure of the quantization unit 105 in thecoding apparatus 1 is different from that of the above-mentionedembodiment and, therefore, only the structure of the quantization unitwill be described hereinafter. In FIG. 5, reference numeral 50061denotes a first auditive weight calculating unit that provides aweighting coefficient to be used by the first stage quantizer 40051 inthe multistage quantization means 4005, on the basis of the spectrum ofthe input audio signal, the auditive sensitivity characteristic showingthe auditive nature of human beings, and the LPC spectral envelope;50062 denotes a second auditive weight calculating unit that provides aweighting coefficient to be used by the second stage quantizer 40052 inthe multistage quantization means 4005, on the basis of the spectrum ofinput audio signal, the auditive sensitivity characteristic showing theauditive nature of human beings, and the LPC spectral envelope; and50063 denotes a third auditive weight calculating unit that provides aweighting coefficient to be used by the N-th stage quantizer 40053 inthe multistage quantization means 4005, on the basis of the spectrum ofinput audio signal, the auditive sensitivity characteristic showing theauditive nature of human beings, and the LPC spectral envelope.

In the audio signal coding apparatus according to the first embodiment,all of the plural stages of vector quantizers in the multistagequantization means 4005 perform quantization using the same weightingcoefficient obtained in the auditive weight calculating unit 4006.However, in the audio signal coding apparatus according to this secondembodiment, the plural stages of vector quantizers in the multistagequantization means 4005 perform quantization using individual weightingcoefficients obtained in the first to third auditive weight calculatingunits 50061, 50062, and 50063, respectively. In this audio signal codingapparatus according to the second embodiment, it is possible to performquantization by weighting according to the frequency weightingcharacteristic obtained in the auditive weighting units 50061 to 50063on the basis of the auditive nature so that an error due to quantizationin each stage of the multistage quantization means 4005 is minimized.For example, a weighting coefficient is calculated on the basis of thespectral envelope in the first auditive weighting unit 50061, aweighting coefficient is calculated on the basis of the minimum audiblelimit characteristic in the second auditive weighting unit 50062, and aweighting coefficient is calculated on the basis of the auditive maskingcharacteristic in the third auditive weighting unit 50063.

As described above, according to the audio signal coding apparatus ofthe second embodiment, since the plural-stages of quantizers 40051 to40053 in the multistage quantization means 4005 perform quantizationusing the individual weighting coefficients obtained in the auditiveweight calculating units 50061 to 50063, respectively, efficientquantization can be performed by effectively utilizing the auditivenature of human beings.

Embodiment 3

FIG. 6 is a block diagram illustrating the structure of an audio signalcoding apparatus according to a third embodiment of the invention. Inthis embodiment, only the structure of the quantization unit 105 in thecoding apparatus 1 is different from that of the above-mentionedembodiment and, therefore, only the structure of the quantization unitwill be described hereinafter. In FIG. 6, reference numeral 60021denotes a first-stage quantization unit that vector-quantizes anormalized MDCT signal; 60023 denotes a second-stage quantization unitthat quantizes a quantization error signal caused by the quantization inthe first-stage quantization unit 60021; and 60022 denotes an auditiveselection means that selects, from the quantization error caused by thequantization in the first-stage quantization unit 60021, a frequencyband of highest importance to be quantized in the second-stagequantization unit 60023, on the basis of the auditive sensitivitycharacteristic.

A description is given of the operation. The normalized MDCTcoefficients are subjected to vector quantization in the first-stagequantization unit 60021. In the auditive selection means 60022, afrequency band, in which an error signal due to the vector quantizationis large, is decided on the basis of the auditive scale, and a blockthereof is extracted. In the second-stage quantization unit 60023, theerror signal of the selected block is subjected to vector quantization.The results obtained in the respective quantization units are output asindices.

FIG. 7 is a block diagram illustrating, in detail, the first and secondstage quantization units and the auditive selection unit, included inthe audio signal coding apparatus shown in FIG. 6. In FIG. 7, referencenumeral 7031 denotes a first vector quantizer that vector-quantizes thenormalized MDCT coefficients; and 70032 denotes an inverse quantizerthat inversely quantizes the quantization result of the first quantizer70031, and a quantization error signal zi due to the quantization by thefirst quantizer 70031 is obtained by obtaining a difference between theoutput from the inverse quantizer 70032 and a residual signal si.Reference numeral 70033 denotes auditive sensitivity characteristic hishowing the auditive nature of human beings, and the minimum audiblelimit characteristic is used here. Reference numeral 70035 denotes aselector that selects a frequency band to be quantized by the secondvector quantizer 70036, from the quantization error signal zi due to thequantization by the first quantizer 70031. Reference numeral 70034denotes a selection scale calculating unit that calculates a selectionscale for the selecting operation of the selector 70035, on the basis ofthe error signal zi, the LPC spectral envelope li, and the auditivesensitivity characteristic hi.

Next, the selecting operation of the auditive selection unit will bedescribed in detail.

In the first vector quantizer 70031, first of all, a residual signal inone frame comprising N pieces of elements is divided into pluralsub-vectors by a vector divider in the first vector quantizer 70031shown in FIG. 8(a), and the respective sub-vectors are subjected tovector quantization by the N pieces of quantizers 1˜N in the firstvector quantizer 70031. The method of vector division and quantizationis as follows. For example, as shown in FIG. 8(b), N pieces of elementsbeing arranged in ascending order of frequency are divided into NSpieces of sub-blocks at equal intervals, and NS pieces of sub-vectorscomprising N/NS pieces of elements, such as a sub-vector comprising onlythe first elements in the respective sub-blocks, a sub-vector comprisingonly the second elements thereof, . . . , are created, and vectorquantization is carried out for each sub-vector. The division number andthe like are decided on the basis of the requested coding rate.

After the vector quantization, the quantized code is inversely quantizedby the inverse quantizer 70032 to obtain a difference from the inputsignal, thereby providing an error signal zi in the first vectorquantizer 70031 as shown in FIG. 9(a).

Next, in the selector 70035, from the error signal Zi, a frequency blockto be quantized more precisely by the second quantizer 70036 is selectedon the basis of the result selected by the selection scale calculatingunit 70034.

In the selection scale calculating unit 70034, using the error signalZi, the LPC spectral envelope li as shown in FIG. 9(b) obtained in theLPC analysis unit, and the auditive sensitivity characteristic hi, foreach element in the frame divided into N elements on the frequency axis,

g=(zi*hi)/hi

is calculated.

As the auditive sensitivity characteristic hi, for example, the minimumaudible limit characteristic shown in FIG. 9(c) is used. This is acharacteristic showing a region that cannot be heard by human beings,obtained experimentally. Therefore, it may be said that l/hi, which isthe inverse number of the auditive sensitivity characteristic hi, showsthe auditive importance of human beings. In addition, it may be saidthat the value g, which is obtained by multiplying the error signal zi,the spectral envelope li, and the inverse number of the auditivesensitivity characteristic hi, shows the importance of precisequantization at the frequency.

FIG. 10 is a block diagram illustrating, in detail, other examples ofthe first and second stage quantization units and the auditive selectionunit, included in the audio signal coding apparatus shown in FIG. 6. InFIG. 10, the same reference numerals as those in FIG. 7 designate thesame or corresponding parts. In the example shown in FIG. 10, theselection scale (importance) g is obtained using the spectral envelopeli and the auditive sensitivity characteristic hi, without using theerror signal zi, by calculating,

g=li/hi

FIG. 11 is a block diagram illustrating, in detail, still other examplesof the first and second stage quantization units and the auditiveselection unit, included in the audio signal coding apparatus shown inFIG. 6. In FIG. 11, the same reference numerals as those shown in FIG. 7designate the same or corresponding parts, and reference numeral 110042denotes a masking amount calculating unit that calculates an amount tobe masked by the auditive masking characteristic, from the spectrum ofthe input audio frequency which has been MDCT-transformed in thetime-to-frequency transform unit.

In the example shown in FIG. 11, the auditive sensitivity characteristichi is obtained frame by frame according to the following manner. Thatis, the masking characteristic is calculated from the frequency spectraldistribution of the input signal, and the minimum audible limitcharacteristic is added to the masking characteristic, thereby to obtainthe auditive sensitivity characteristic hi of the frame. The operationof the selection scale calculating unit 70034 is identical to thatdescribed with respect to FIG. 10.

FIG. 12 is a block diagram illustrating, in detail, still other examplesof the first and second stage quantization units and the auditiveselection unit, included in the audio signal coding apparatus shown inFIG. 6. In FIG. 12, the same reference numerals as those shown in FIG. 7designate the same or corresponding parts, and reference numeral 120043denotes a masking amount correction unit that corrects the maskingcharacteristic obtained in the masking amount calculating unit 110042,using the spectral envelope li, the residual signal si, and the errorsignal zi.

In the example shown in figure 12, the auditive sensitivitycharacteristic hi is obtained frame by frame in the following manner.Initially, the masking characteristic is calculated from the frequencyspectral distribution of the input signal in the masking amountcalculating unit 110042. Next, in the masking amount correction unit120043, the calculated masking characteristic is corrected according tothe spectral, envelope li, the residual signal si, and the error signalzi. The audio sensitivity characteristic hi of the frame is obtained byadding the minimum audible limit characteristic to the corrected maskingcharacteristic. An example of a method of correcting the maskingcharacteristic will be described hereinafter.

Initially, a frequency (fm) at which the characteristic of maskingamount Mi, which has already been calculated, attains the maximum valueis obtained. Next, how precisely the signal having the frequency fm isreproduced is obtained from the spectral intensity of the frequency fmat the input and the size of the quantization error spectrum. Forexample,

γ=1−(gain of quantization error of fm)/(gain of fm at input)

When the value of γ is close to 1, it is not necessary to transform themasking characteristic already obtained. However, when it is close to 0,the masking characteristic is corrected so as to be decreased. Forexample, the masking characteristic can be corrected by transforming itby raising it to a higher power with the coefficient γ, as follows.

hi=Mi ^(γ)  (31)

Next, a description is given of the operation of the selector 70035.

In the selector 70035, each of continuous elements in a frame ismultiplied by a window (length W), and a frequency block in which avalue G obtained by accumulating the values of importance g within thewindow attains the maximum is selected. FIG. 13 is a diagram showing anexample where a frequency block (length W) of highest importance isselected. For simplification, the length of the window should be set atinteger multiples of N/NS (FIG. 13 shows one which is not an integermultiple.) While shifting the window by N/NS pieces, the accumulatedvalue G of the importance g within the window frame is calculated, and afrequency block having a length W that gives the maximum value of G isselected.

In the second vector quantizer 70032, the selected block in the windowframe is subjected to vector quantization. Although the operation of thesecond vector quantizer 70032 is identical to that of the first vectorquantizer 70031, since only the frequency block selected by the selector70035 from the error signal zi is quantized as described above, thenumber of elements in the frame to be vector-quantized is small.

Finally, in the case of using the code of the spectral envelopecoefficient, the codes corresponding to the quantization results of therespective vector quantizers, and the selection scale g obtained in anyof the structures shown in FIGS. 7, 11 and 12, information showing fromwhich element does the block selected by the selector 70035 start, isoutput as an index.

On the other hand, in the case of using the selection scale g obtainedin the structure shown in FIG. 10, since only the spectral envelope liand the auditive sensitivity characteristic hi are used, theinformation, i.e., from which element does the selected block start, canbe obtained from the code of the spectral envelope coefficient and thepreviously known auditive sensitivity characteristic hi when inversequantization is carried out. Therefore, it is not necessary to outputthe information relating to the block selection as an index, resultingin an advantage with respect of compressibility.

As described above, according to the audio signal coding apparatus ofthe third embodiment, on the basis of the spectrum of the input audiosignal and the auditive sensitivity characteristic showing the auditivenature of human beings, a frequency block of highest importance forquantization is selected from the frequency blocks of quantization errorcomponent in the first vector quantizer, and the quantization errorcomponent of the first quantizer is quantized with respect to theselected block in the second vector quantizer, whereby efficientquantization can be performed utilizing the auditive nature of humanbeings. Further, in the structures shown in FIGS. 7, 11 and 12, when thefrequency block of highest importance for quantization is selected, theimportance is calculated on the basis of the quantization error in thefirst vector quantizer. Therefore, it is avoided that a portionfavorably quantized in the first vector quantizer is quantized again andan error is generated inversely, whereby quantization maintaining highquality is performed.

Further, when the importance g is obtained in the structure shown inFIG. 10, as compared with the case of obtaining the importance g in thestructure shown in any of FIGS. 7, 11 and 12, the number of indices tobe output is decreased, resulting in increased compression ratio.

In this third embodiment, the quantization unit has the two-stagestructure comprising the first-stage quantization unit 60021 and thesecond-stage quantization unit 60023, and the auditive selection means60022 is disposed between the first-stage quantization unit 60021 andthe second-stage quantization unit 60023. However, the quantization unitmay have a multiple-stage structure of three or more stages and theauditive selection means may be disposed between the respectivequantization units. Also in this structure, as in the third embodimentmentioned above, efficient quantization can be performed utilizing theauditive nature of human beings.

Embodiment 4

FIG. 14 is a block diagram illustrating a structure of an audio signalcoding apparatus according to a fourth embodiment of the presentinvention. In this embodiment, only the structure of the quantizationunit 105 in the coding apparatus 1 is different from that of theabove-mentioned embodiment and, therefore, only the structure of thequantization unit will be described hereinafter. In the figure,reference numeral 140011 denotes a first-stage quantizer thatvector-quantizes the MDCT signal si output from the normalization unit104, using the spectral envelope value li as a weight coefficient.Reference numeral 140012 denotes an inverse quantizer that inverselyquantizes the quantization result of the first-stage quantizer 140011,and a quantization error signal zi of the quantization by thefirst-stage quantizer 140011 is obtained by taking a difference betweenthe output of this inverse quantizer 140012 and a residual signal outputfrom the normalization unit 104. Reference numeral 140013 denotes asecond-stage quantizer that vector-quantizes the quantization errorsignal zi of the quantization by the first-stage quantizer 140011 using,as a weight coefficient, the calculation result obtained in a weightcalculating unit 140017 described later. Reference numeral 140014denotes an inverse quantizer that inversely quantizes the quantizationresult of the second-stage quantizer 140013, and a quantization errorsignal z2i of the quantization by the second-stage quantizer 140013 isobtained by taking a difference between the output of this inversequantizer 140014 and the quantization error signal of the quantizationby the first-stage quantizer 140011. Reference numeral 140015 denotes athird-stage quantizer that vector-quantizes the quantization errorsignal z2i of the quantization by the second-stage quantizer 140013using, as a weight coefficient, the calculation result obtained in theauditive weight calculating unit 4006. Reference numeral 140016 denotesa correlation calculating unit that calculates a correlation between thequantization error signal zi of the quantization by the first-stagequantizer 140011 and the spectral envelope value li. Reference numeral140017 denotes a weight calculating unit that calculates the weightingcoefficient used in the quantization by the second-stage quantizer140013.

A description is given of the operation. In the audio signal codingapparatus according to this fourth embodiment, three stages ofquantizers are employed, and vector quantization is carried out usingdifferent weights in the respective quantizers.

Initially, in the first-stage quantizer 140013, the input residualsignal si is subjected to vector quantization using, as a weightcoefficient, the LPC spectral envelope value li obtained in the outlinequantization unit 302. Thereby, a portion in which the spectral energyis large (concentrated) is subjected to weighting, resulting in aneffect that an auditively important portion is quantized with higherefficiency. As the first-stage vector quantizer 140013, for example, aquantizer identical to the first vector quantizer 70031 according to thethird embodiment may be used.

The quantization result is inversely quantized in the inverse quantizer140012 and, from a difference between this and the input residual signalsi, an error signal zi due to the quantization is obtained.

This error signal zi is further vector-quantized by the second-stagequantizer 140013. Here, on the basis of the correlation between the LPCspectral envelope li and the error signal zi, a weight coefficient iscalculated by the correlation calculating unit 140016 and the weightcalculating unit 140017.

To be specific, in the correlation calculating unit 140016,

α=(Σli*zi)/(Σli*li)

is calculated. This α takes a value in 0<α<1 and shows the correlationbetween them. When α is close to 0, it shows that the first-stagequantization has been carried out precisely on the basis of theweighting of the spectral envelope. When a is close to 1, it shows thatquantization has not been precisely carried out yet. So, using this α,as a coefficient for adjusting the weighting degree of the spectralenvelope li,

li^(α)  (32)

is obtained, and this is used as a weighting coefficient for vectorquantization. The quantization precision is improved by performingweighting again using the spectral envelope according to the precisionof the first-stage quantization and then performing quantization asmentioned above.

The quantization result by the second-stage quantizer 140013 isinversely quantized in the inverse quantizer 140014 in similar manner,and an error signal z2i is extracted, and this error signal z2i isvector-quantized by the third-stage quantizer 140015. The auditiveweight coefficient at this time is calculated by the weight calculator140019 in the auditive weighting calculating unit 4006. For example,using the error signal z2i, the LPC spectral envelope li, and theresidual signal si,

N=Σz 2 i*li

S=Σsi*li

β=1−(N/S)

are obtained.

On the other hand, in the auditive masking calculator 140018 in theauditive weighting calculating unit 4006, the auditive maskingcharacteristic mi is calculated according to, for example, an auditivemodel used in an MPEG audio standard method. This is overlapped with theabove-described minimum audible limit characteristic hi to obtain thefinal masking characteristic Mi.

Then, the final masking characteristic Mi is raised to a higher powerusing the coefficient β calculated in the weight calculating unit140019, and the inverse number of this value is multiplied by l toobtain

l/Mi^(β)  (33)

and this is used as a weight coefficient for the third-stage vectorquantization.

As described above, in the audio signal coding apparatus according tothis fourth embodiment, the plural quantizers 140011, 140013, and 140015perform quantization using different weighting coefficients, includingweighting in view of the auditive sensitivity characteristic, wherebyefficient quantization can be performed by effectively utilizing theauditive nature of human beings.

Embodiment 5

FIG. 15 is a block diagram illustrating the structure of an audio signalcoding apparatus according to a fifth embodiment of the presentinvention.

The audio signal coding apparatus according to this fifth embodiment isa combination of the third embodiment shown in FIG. 6 and the firstembodiment shown in FIG. 4 and, in the audio signal coding apparatusaccording to the third embodiment shown in FIG. 6, a weightingcoefficient, which is obtained by using the auditive sensitivitycharacteristic in the auditive weighting calculating unit 4006, is usedwhen quantization is carried out in each quantization unit. Since theaudio signal coding apparatus according to this fifth embodiment is soconstructed, both of the effects provided by the first embodiment andthe third embodiment are obtained.

Further, likewise, the third embodiment shown in FIG. 6 may be combinedwith the structure according to the second embodiment or the fourthembodiment, and an audio signal coding apparatus obtained by eachcombination can provide both of the effects provided by the secondembodiment and the third embodiment or both of the effects provided bythe fourth embodiment and the third embodiment.

While in the aforementioned first to fifth embodiments the multistagequantization unit has two or three stages of quantization units, it isneedless to say that the number of stages of the quantization unit maybe four or more.

Furthermore, the order of the weight coefficients used for vectorquantization in the respective stages of the multistage quantizationunit is not restricted to that described for the aforementionedembodiments. For example, the weighting coefficient in view of theauditive sensitivity characteristic may be used in the first stage, andthe LPC spectral envelope may be used in and after the second stage.

Embodiment 6

FIG. 16 is a block diagram illustrating an audio signal coding apparatusaccording to a sixth embodiment of the present invention. In thisembodiment, since only the structure of the quantization unit 105 in thecoding apparatus 1 is different from that of the above-mentionedembodiment, only the structure of the quantization unit will bedescribed hereinafter.

In FIG. 16, reference numeral 401 denotes a first sub-quantization unit401, 402 denotes a second sub-quantization unit that receives an outputfrom the first sub-quantization unit 401, and 403 denotes a thirdsub-quantization unit that receives the output from the secondsub-quantization unit 402.

Next, a description is given of the operation of the quantization unit105. A signal input to the first sub-quantization unit 401 is the outputfrom the normalization unit 104 of the coding apparatus, i.e.,normalized MDCT coefficients. However, in the structure having nonormalization unit 104, it is the output from the MDCT unit 103. In thefirst sub-quantization unit 401, the input MDCT coefficients aresubjected to scalar quantization or vector quantization, and indicesexpressing the parameters used for the quantization are encoded.Further, quantization errors with respect to the input MDCT coefficientsdue to the quantization are calculated, and they are output to thesecond sub-quantization unit 402. In the first sub-quantization unit401, all of the MDCT coefficients may be quantized, or only a portion ofthem may be quantized. Of course, when only a portion thereof isquantized, quantization errors in the bands which are not quantized bythe first sub-quantization unit 401 will become input MDCT coefficientsof the not-quantized bands.

Next, the second sub-quantization unit 402 receives the quantizationerrors of the MDCT coefficients obtained in the first sub-quantizationunit 401 and quantizes them. For this quantization, like the firstsub-quantization unit 401, scalar quantization or vector quantizationmay be used. The second sub-quantization unit 402 codes the parametersused for the quantization as indices. Further, it calculatesquantization errors due to the quantization, and outputs them to thethird sub-quantization unit 403. This third sub-quantization unit 403 isidentical in structure to the second sub-quantization unit.

The numbers of MDCT coefficients, i.e., band widths, to be quantized bythe first sub-quantization unit 401, the second sub-quantization unit402, and the third sub-quantization unit 403 are not necessarily equalto each other, and the bands to be quantized are not necessarily thesame. Considering the auditive characteristic of human beings, it isdesired that both of the second sub-quantization unit 402 and the thirdsub-quantization unit 403 are set so as to quantize the band of the MDCTcoefficients showing the low-frequency component.

As described above, according to the sixth embodiment of the invention,when quantization is performed, the quantization unit is provided instages, and the band width to be quantized by the quantization unit isvaried between the adjacent stages, whereby coefficients in an arbitraryband among the input MDCT coefficients, for example, coefficientscorresponding to the low-frequency component which is auditivelyimportant for human beings, are quantized. Therefore, even when an audiosignal is coded at a low bit rate, i.e., a high compression ratio, it ispossible to perform high-definition audio reproduction at the receivingend.

Embodiment 7

Next, an audio signal coding apparatus according to a seventh embodimentof the invention will be described using FIG. 17. In this embodiment,since only the structure of the quantization unit 105 in the codingapparatus 1 is different from that of the above-mentioned embodiment,only the structure of the quantization unit will be explained. In FIG.17, reference numeral 501 denotes a first sub-quantization unit (vectorquantizer), 502 denotes a second sub-quantization unit, and 503 denotesa third sub-quantization unit. This seventh embodiment is different instructure from the sixth embodiment in that the first quantization unit501 divides the input MDCT coefficients into three bands and quantizesthe respective bands independently. Generally, when quantization iscarried out using a method of vector quantization, vectors areconstituted by extracting some elements from input MDCT coefficients,whereby vector quantization is performed. In the first sub-quantizationunit 501 according to this seventh embodiment, when creating vectors byextracting some elements from the input MDCT coefficients, quantizationof the low band is performed using only the elements in the low band,quantization of the intermediate band is performed using only theelements in the intermediate band, and quantization of the high band isperformed using only the elements in the high band, whereby therespective bands are subjected to vector quantization. The firstsub-quantization unit 501 is seemed to be composed of three-dividedvector quantizers.

Although in this seventh embodiment, a method of dividing the band to bequantized into three bands, i.e., low band, intermediate band, and highband, is described as an example, the number of divided bands may beother than three. Further, with respect to the second sub-quantizationunit 502 and the third sub-quantization unit 503, as well as the firstquantization unit 501, the band to be quantized may be divided intoseveral bands.

As described above, according to the seventh embodiment of theinvention, when quantization is carried out, the input MDCT coefficientsare divided into three bands and quantized independently, so that theprocess of quantizing the auditively important band with priority can beperformed in the first-time quantization. Further, in the subsequentquantization units 502 and 503, the MDCT coefficients in this band aresubjected to further quantization by stages, whereby the quantizationerror is reduced furthermore, and higher-definition audio reproductionis realized at the receiving end.

Embodiment 8

An audio signal coding apparatus according to an eighth embodiment ofthe invention will be described using FIG. 18. In this eighthembodiment, since only the structure of the quantization unit 105 in thecoding apparatus 1 is different from that of the above-mentioned firstembodiment, only the structure of the quantization unit will beexplained. In FIG. 18, reference numeral 601 denotes a firstsub-quantization unit, 602 denotes a first quantization band selectionunit, 603 denotes a second sub-quantization unit, 604 denotes a secondquantization band selection unit, and 605 denotes a thirdsub-quantization unit. This eighth embodiment is different in structurefrom the sixth and seventh embodiments in that the first quantizationband selection unit 602 and the second quantization band selection unit604 are added.

Hereinafter, the operation will be described. The first quantizationband selection unit 602 calculates a band, of which MDCT coefficientsare to be quantized by the second sub-quantization unit 602, using thequantization error output from the first sub-quantization unit 601.

For example, j which maximizes esum(j) given in formula (10) iscalculated, and a band ranging from $\begin{matrix}{{{esum}\quad (j)} = {\sum\limits_{i = {j \cdot {OFFSET}}}^{{j \cdot {OFFSET}} + {BANDWIDTH}}\quad {{fdct}_{err}\quad (i)^{2}}}} & (10)\end{matrix}$

where OFFSET is the constant, and BANDWIDTH is the total samplecorresponding to a band width to be quantized by the secondsub-quantization unit 603. The first quantization band selection unit602 codes, for example, the j which gives the maximum value in formula(10), as an index. The second sub-quantization unit 603 quantizes theband selected by the first quantization band selection unit 602. Thesecond quantization band selection unit 604 is implemented by the samestructure as the first selection unit except that its input is thequantization error output from the second sub-quantization unit 603, andthe band selected by the second quantization band selection unit 604 isinput to the third sub-quantization unit 605.

Although in the first quantization band selection unit 602 and thesecond quantization band selection unit 604, a band to be quantized bythe next quantization unit is selected using formula (10), it may becalculated using a value obtained by multiplying a value used fornormalization by the normalization unit 104 and a value in view of theauditive sensitivity characteristic of human beings relative tofrequencies, as shown in formula (11). $\begin{matrix}{{{esum}\quad (j)} = {\sum\limits_{i = {j \cdot {OFFSET}}}^{{j \cdot {OFFSET}} + {BANDWIDTH}}\quad \left\{ {{fdct}_{err}\quad {(i) \cdot {env}}\quad {(i) \cdot {zxc}}\quad (i)} \right\}^{2}}} & (11)\end{matrix}$

where env(i) is obtained by dividing the output from the MDCT unit 103with the output from the normalization unit 104, and zxc(i) is the tablein view of the auditive sensitivity characteristic of human beingsrelative to frequencies, and an example thereof is shown in Graph 2. Informula (11), zxc(i) may be always 1 so that it is not considered.

Further, it is not necessary to provide plural stages of quantizationband selection units, i.e., only the first quantization band selectionunit 602 or the second quantization band selection unit 604 may be used.

As described above, according to the eighth embodiment, whenquantization is performed in plural stages, a quantization bandselection unit is disposed between adjacent stages of quantization unitsto make the band to be quantized variable. Thereby, the band to bequantized can be varied according to the input signal, and the degree offreedom in the quantization is increased.

Hereinafter, a description is given of the detailed operation by aquantization method of the quantization unit included in the codingapparatus 1 according to any of the first to eighth embodiments, usingFIG. 1 and FIG. 19. From the normalized MDCT coefficients 1401 input toeach sub-quantization unit, some of them are extracted according to arule to constitute sound source sub-vectors 1403. Likewise, assumingthat the coefficient streams, which are obtained by dividing the MDCTcoefficients to be input to the normalization unit 104 with the MDCTcoefficients 1401 normalized by the normalization unit 104, arenormalized components 1402, some of these components are extractedaccording to the same rule as that for extracting the sound sourcesub-vectors from the MDCT coefficients 1401, thereby to constituteweight sub-vectors 1404. The rule for extracting the sound sourcesub-vectors 1403 and the weight sub-vectors 1404 from the MDCTcoefficients 1401 and the normalized components 1402, respectively, isshown in, for example, formula (14). $\begin{matrix}{{{subvector}_{i}\quad (j)} = \left( \begin{matrix}{{vector}\quad \left( {{\frac{VTOTAL}{CR} \cdot i} + j} \right)} & \quad \\0 & \begin{matrix}{{{\frac{VTOTAL}{CR} \cdot i} + j} < {TOTAL}} \\{{{\frac{VTOTAL}{CR} \cdot i} + j} \geq {TOTAL}}\end{matrix}\end{matrix} \right.} & (14)\end{matrix}$

where the j-th element of the i-th sound source sub-vector issubvector_(i)(j), the MDCT coefficients are vector( ), the total elementnumber of the MDCT coefficients 1401 is TOTAL, the element number of thesound source sub-vectors 1403 is CR, and VTOTAL is set to a value equalto or larger than TOTAL and VTOTAL/CR should be an integer. For example,when TOTAL is 2048, CR=19 and VTOTAL=2052, or CR=23 and VTOTAL=2070, orCR=21 and VTOTAL=2079. The weight sub-vectors 1404 can be extracted bythe procedure of formula (14). The vector quantizer 1405 selects, fromthe code vectors in the code book 1409, a code vector having a minimumdistance between it and the sound source sub-vector 1403, after beingweighted by the weight sub-vector 1404. Then, the quantizer 1405 outputsthe index of the code vector having the minimum distance, and a residualsub-vector 1404 which corresponds to the quantization error between thecode vector having the minimum distance and the input sound sourcesub-vector 1403. An example of actual calculation procedure will bedescribed on the premise that the vector quantizer 1405 is composed ofthree constituents: a distance calculating means 1406, a code decisionmeans 1407, and a residual generating means 1408. The distancecalculating means 1406 calculates the distance between the i-th soundsource sub-vector 1403 and the k-th code vector in the code book 1409using, for example, formula (15). $\begin{matrix}{{dik} = {\sum\limits_{j = 0}^{{CR} - 1}\quad {w_{j}^{R}\quad \left( {{{subvector}_{i}\quad (j)} - {C_{k}\quad (j)}} \right)^{S}}}} & (15)\end{matrix}$

where wj is the j-th element of the weight sub-vector, ck(j) is the j-thelement of the k-th code vector, R and S are norms for distancecalculation, and the values of R and S are desired to be 1, 1.5, 2.These norms R and S may have different values. Further, dik is thedistance of the k-th code vector from the i-th sound source sub-vector.The code decision means 1407 selects a code vector having a minimumdistance among the distances calculated by formula (15) or the like, andcodes the index thereof. For example, when diu is the minimum value, theindex to be coded for the i-th sub-vector is u. The residual generatingmeans 1408 generates residual sub-vectors 1410 using the code vectorsselected by the code decision means 1407, according to formula (16).

res_(i)(j)=subvector_(i)(j)−C _(u)(j)  (16)

wherein the j-th element of the i-th residual sub-vector 1410 isresi(j), and the j-th element of the code vector selected by the codedecision means 1407 is cu(j). The residual sub-vectors 1410 are retainedas MDCT coefficients to be quantized by the subsequent sub-quantizationunits, by executing the inverse process of formula (14) or the like.However, when a band being quantized does not influence on thesubsequent sub-quantization units, i.e., when the subsequentsub-quantization units are not required to perform quantization, theresidual generating means 1408, the residual sub-vectors 1410, and thegeneration of the MDCT 1411 are not necessary. Although the number ofcode vectors possessed by the code book 1409 is not specified, when thememory capacity, calculating time and the like are considered, thenumber is desired to be about 64.

As another embodiment of the vector quantizer 1405, the followingstructure is available. That is, the distance calculating means 1406calculates the distance using formula (17). $\begin{matrix}{{dik} = \left\{ \begin{matrix}{\sum\limits_{j = 0}^{{CR} - 1}\quad {w_{j}^{R}\quad \left( {{{subvector}_{i}\quad (j)} - {C_{k}\quad (j)}} \right)^{S}}} & {k < K} \\{\sum\limits_{j = 0}^{{CR} - 1}\quad {w_{j}^{R}\quad \left( {{{subvector}_{i}\quad (j)} - {C_{K - k}\quad (j)}} \right)^{S}}} & {k \geq K}\end{matrix} \right.} & (17)\end{matrix}$

wherein K is the total number of code vectors used for the coderetrieval of the code book 1409.

The code decision means 1407 selects k that gives a minimum value of thedistance dik calculated in formula (17), and codes the index thereof.Here, k is a value in a range from 0 to 2K−1. The residual generatingmeans 1408 generates the residual sub-vectors 1410 using formula (18).$\begin{matrix}{{{resi}(j)} = \left\{ \begin{matrix}{{{subvector}_{i}(j)} - {C_{u}(j)}} & {0 \leq k < K} \\{{{subvector}_{i}(j)} + {C_{u}(j)}} & {K \leq k < {2\quad K}}\end{matrix} \right.} & (18)\end{matrix}$

Although the number of code vectors possessed by the code book 1409 isnot restricted, when the memory capacity, calculation time and the likeare considered, it is desired to be about 64.

Further, although the weight sub-vectors 1404 are generated from thenormalized components 1402, it is possible to generate weightsub-vectors by multiplying the weight sub-vectors 1404 by a weight inview of the auditive characteristic of human beings.

Embodiment 9

Next, an audio signal decoding apparatus according to a ninth embodimentof the present invention will be described using FIGS. 20 to 24. Theindices output from the coding apparatus 1 are divided broadly into theindices output from the normalization unit 104 and the indices outputfrom the quantization unit 105. The indices output from thenormalization unit 104 are decoded by the inverse normalization unit107, and the indices output from the quantization unit 105 are decodedby the inverse quantization unit 106. The inverse quantization unit 106can perform decoding using only a portion of the indices output from thequantization unit 105.

That is, assuming that the quantization unit 105 has the structure shownin FIG. 17, a description is given of the case where inversequantization is carried out using the inverse quantization unit havingthe structure of FIG. 20. In FIG. 20, reference numeral 701 designates afirst low-band-component inverse quantization unit. The firstlow-band-component inverse quantization unit 701 performs decoding usingonly the indices of the low-band components of the first sub-quantizer501.

Thereby, regardless of the quantity of data transmitted from the codingapparatus 1, an arbitrary quantity of data of the coded audio signal canbe decoded, whereby the quantity of data coded can be different from thequantity of data decoded. Therefore, the quantity of data to be decodedcan be varied according to the communication environment on thereceiving end, and high-definition sound quality can be obtained stablyeven when an ordinary public telephone network is used.

FIG. 21 is a diagram showing the structure of the inverse quantizationunit included in the audio signal decoding apparatus, which is employedwhen inverse quantization is carried out in two stages. In FIG. 21,reference numeral 704 denotes a second inverse quantization unit. Thissecond inverse quantization unit 704 performs decoding using the indicesfrom the second sub-quantization unit 502. Accordingly, the output fromthe first low-band-component inverse quantization unit 701 and theoutput from the second inverse quantization unit 704 are added and theirsum is output from the inverse quantization unit 106. This addition isperformed to the same band as the band quantized by eachsub-quantization unit in the quantization.

As described above, the indices from the first sub-quantization unit(low-band) are decoded by the first low-band-component inversequantization unit 701 and, when the indices from the secondsub-quantization unit are inversely quantized, the output from the firstlow-band-component inverse quantization unit 701 is added thereto,whereby the inverse quantization is carried out in two stages.Therefore, the audio signal quantized in multiple stages can be decodedaccurately, resulting in a higher sound quality.

Further, FIG. 22 is a diagram illustrating the structure of the inversequantization unit included in the audio signal decoding apparatus, inwhich the object band to be processed is extended when the two-stageinverse quantization is carried out. In FIG. 22, reference numeral 702denotes a first intermediate-band-component inverse quantization unit.This first intermediate-band-component inverse quantization unit 702performs decoding using the indices of the intermediate-band componentsfrom the first sub-quantization unit 501. Accordingly, the output fromthe first low-band-component inverse quantization unit 701, the outputfrom the second inverse quantization unit 704, and the output from thefirst intermediate-band-component inverse quantization unit 702 areadded and their sum is output from the inverse quantization unit 106.This addition is performed to the same band as the band quantized byeach sub-quantization unit in the quantization. Thereby, the band of thereproduced sound is extended, and an audio signal of higher quality isreproduced.

Further, FIG. 23 is a diagram showing the structure of the inversequantization unit included in the audio signal decoding apparatus, inwhich inverse quantization is carried out in three stages by the inversequantization unit having the structure of FIG. 22. In FIG. 23, referencenumeral 705 denotes a third inverse quantization unit. The third inversequantization unit 705 performs decoding using the indices from the thirdsub-quantization unit 503. Accordingly, the output from the firstlow-band-component inverse quantization unit 701, the output from thesecond inverse quantization unit 704, the output from the firstintermediate-band-component inverse quantization unit 702, and theoutput from the third inverse quantization unit 705 are added and theirsum is output from the inverse quantization unit 106. This addition isperformed to the same band as the band quantized by eachsub-quantization unit in the quantization.

Further, FIG. 24 is a diagram illustrating the structure of the inversequantization unit included in the audio signal decoding apparatus, inwhich the object band to be processed is extended when the three-stageinverse quantization is carried out in the inverse quantization unithaving the structure of FIG. 23. In FIG. 24, reference numeral 703denotes a first high-band-component inverse quantization unit. Thisfirst high-band-component inverse quantization unit 703 performsdecoding using the indices of the high-band components from the firstsub-quantization unit 501. Accordingly, the output from the firstlow-band-component inverse quantization unit 701, the output from thesecond inverse quantization unit 704, the output from the firstintermediate-band-component inverse quantization unit 702, the outputfrom the third inverse quantization unit.705, and the output from thefirst high-band-component inverse quantization unit 703 are added andtheir sum is output from the inverse quantization unit 106. Thisaddition is performed to the same band as the band quantized by eachsub-quantization unit in the quantization.

While this ninth embodiment is described for the case where the decodingunit 106 inversely decodes the data quantized by the quantization unit105 having the structure of FIG. 7, similar inverse quantization can becarried out even when the quantization unit 105 has the structure shownin FIG. 16 or 18.

Furthermore, when coding is carried out using the quantization unithaving the structure shown in FIG. 17 and decoding is carried out usingthe inverse quantization unit having the structure shown in FIG. 24, asshown in FIG. 25, after the low-band indices from the firstsub-quantization unit are inversely quantized, the indices from thesecond sub-quantization unit 502 in the next stage are inverselyquantized, and the intermediate-band indices from the firstsub-quantization unit are inversely quantized. In this way, the inversequantization to extend the band and the inverse quantization to reducethe quantization error are alternatingly repeated. However, when asignal coded by the quantization unit having the structure shown in FIG.16 is decoded using the inverse quantization unit having the structureshown in FIG. 24, since there is no divided bands, the quantizedcoefficients are successively decoded by the inverse quantization unitin the next stage.

A description is given of the detailed operation of the inversequantization unit 107 as a constituent of the audio signal decodingapparatus 2, using FIG. 1 and FIG. 26.

For example, the inverse quantization unit 107 is composed of the firstlow-band inverse quantization unit 701 when it has the inversequantization unit shown in FIG. 20, and it is composed of two inversequantization units, i.e., the first low-band inverse quantization unit701 and the second inverse quantization unit 704, when it has theinverse quantization unit shown in FIG. 21.

The vector inverse quantizer 1501 reproduces the MDCT coefficients usingthe indices from the vector quantization unit 105. When thesub-quantization unit has the structure shown in FIG. 20, inversequantization is carried out as follows. An index number is decoded, anda code vector having the number is selected from the code book 1502. Itis assumed that the content of the code book 1502 is identical to thatof the code book of the coding apparatus. The selected code vectorbecomes, as a reproduced vector 1503, an MDCT coefficient 1504 inverselyquantized by the inverse process of formula (14).

When the sub-quantization unit has the structure shown in FIG. 21,inverse quantization is carried out as follows. An index number k isdecoded, and a code vector having the number u calculated in formula(19) is selected from the code book 1502. $\begin{matrix}{u = \left\{ \begin{matrix}k & {0 \leq k < K} \\{k - K} & {K \leq k < {2\quad K}}\end{matrix} \right.} & (19)\end{matrix}$

A reproduced sub-vector is generated using formula (20). $\begin{matrix}{{{resi}(j)} = \left\{ \begin{matrix}{C_{u}{l(j)}} & {u = k} \\{- {C_{u}(j)}} & {u \neq k}\end{matrix} \right.} & (20)\end{matrix}$

wherein the j-th element of the i-th reproduced sub-vector is resi(j).

Next, a description is given of the detailed structure of the inversenormalization unit 107 as a constituent of the audio signal decodingapparatus B2, using FIG. 1 and FIG. 27. In FIG. 27, reference numeral1201 denotes a frequency outline inverse quantization unit, 1202 denotesa band amplitude inverse normalization unit, and 1203 denotes a bandtable. The frequency outline inverse normalization unit 1201 receivesthe indices from the frequency outline normalization unit 1201,reproduces the frequency outline, and multiplies the output from theinverse quantization unit 106 by the frequency outline. The bandamplitude inverse normalization unit 1202 receives the indices from theband amplitude normalization unit 202, and restores the amplitude ofeach band shown in the band table 1203, by multiplication. Assuming thatthe value of each band restored using the indices from the bandamplitude normalization unit B202 is qavej, the operation of the bandamplitude inverse normalization unit 1202 is given by formula (12).

dct(i)=n _(—) dct(i)·gave_(j) bjlow≦i≦bjhigh  (12)

wherein the output from the frequency outline inverse normalization unit1201 is n_dct(i), and the output from the band amplitude inversenormalization unit 1202 is dct(i). In addition, the band table 1203 andthe band table 203 are identical.

Next, a description is given of the detailed structure of the frequencyoutline inverse normalization unit 1201 as a constituent of the audiosignal decoding apparatus 2, using FIG. 28. In FIG. 28, referencenumeral 1301 designates an outline inverse quantization unit, and 1302denotes an envelope characteristic inverse quantization unit. Theoutline inverse quantization unit 1301 restores parameters showing thefrequency outline, for example, linear prediction coefficients, usingthe indices from the outline quantization unit 301 in the codingapparatus. When the restored coefficients are linear predictioncoefficients, the quantized envelope characteristics are restored bycalculating them similarly in formula (8). When the restoredcoefficients are not linear prediction coefficients, for example, whenthey are LSP coefficients, the envelope characteristics are restored bytransforming them to frequency characteristics. The envelopecharacteristic inverse quantization unit 1302 multiplies the restoredenvelope characteristics by the output from the inverse quantizationunit 106 as shown in formula (13), and outputs the result.

mdct(i)=fdct(i)·env(i)  (13)

Embodiment 10

Hereinafter, an audio signal coding apparatus according to a tenthembodiment of the present invention will be described with reference tothe drawings. FIG. 29 is a diagram illustrating the detailed structureof an audio signal coding apparatus according to the tenth embodiment.In the figure, reference numeral 29003 denotes a transmission-side codebook having a plurality of audio codes which are representative valuesof feature amounts of audio signal, 2900102 denotes an audio codeselection unit, and 2900107 denotes a phase information extraction unit.

Hereinafter, a description is given of the operation.

Although MDCT coefficients are regarded as an input signal in this case,DFT (discrete Fourier transform) coefficients or the like may be used aslong as it is a time-to-frequency transformed signal.

As shown in FIG. 30, when data on the frequency axis is regarded as onesound source vector, some elements are extracted from the sound sourcevector to form a sub-vector. When this sub-vector is regarded as theinput vector shown in FIG. 29, the audio code selection unit 2900102calculates distances between the input vector and the respective codesin the transmission-side code book 29003, selects a code having aminimum distance, and outputs the code index of the selected code in thetransmission-side code book 29003.

A description is given of the detailed operation of the coding apparatususing FIG. 29 and FIG. 31. It is assumed that coding is carried out with10 bits because it is intended for 20 KHz. Further, in the phaseinformation extraction unit 2900107, phases are extracted from twoelements on the low-frequency side, i.e., 2 bits. The input to the audiocode selection unit 2900102 is a sub-vector obtained as follows. Whencoefficients obtained by MDCT are regarded as one vector, this vector isdivided into plural sub-vectors so that each sub-vector is composed ofsome elements, for example, about 20 elements. In this case, thesub-vector is expressed by X0˜X19, and a sub-vector element, of whichthe number appended to X is smaller, corresponds to an MDCT coefficienthaving a lower frequency component. The low frequency component isauditively important information for human beings and, therefore, toperform coding of these elements with priority results in that thedegradation in sound quality is hardly sensed by human beings when beingreproduced.

The audio code selection unit 2900102 calculates distances between thefeature vector and the respective codes in the transmission-side codebook 29003. For example, when the code index is i, the distance Di of acode having the code index i is calculated in formula (21).$\begin{matrix}{D_{i} = {{\sum\limits_{i = 0}^{N}\quad {\sum\limits_{j = 0}^{M}\quad \left\{ {{{abs}\quad ({Cij})} - {{abs}\quad ({Xj})}} \right\}^{P}}} + {\sum\limits_{i = 0}^{N}\quad {\sum\limits_{j = {M + 1}}^{19}\quad \left\{ {{Cij} - {Xj}} \right\}^{P}}}}} & (21)\end{matrix}$

where N is the number of all codes in the transmission-side code book29003, Cij is the value of the j-th element in code index I. In thistenth embodiment, M is a number smaller than 19, for example, 1. P isthe norm for distance calculation and, for example, it is 2. Further,abs( ) means absolute calculation.

The phase information extraction unit 2900107 outputs the coded index igiving a minimum distance Di, and M pieces of phase information Ph(j)j=0 to M. The phase information Ph(j) is expressed by formula (22).$\begin{matrix}{{{Ph}(j)} = \left\{ \begin{matrix}{{1\quad {at}\quad {Cji}*{Xj}} \geq 0} \\{{{- 1}\quad {at}\quad {Cji}*{Xj}} < 0}\end{matrix} \right.} & (22)\end{matrix}$

When the input vector is a sub-vector of a vector obtained by subjectingan audio signal to MDCT, generally, the auditive importance of thecoefficient is higher as the appended character j of Xj is smaller. So,in this structure, with respect to the phases (negative or positive)corresponding to the elements of the low-frequency components of eachsub-vector, these data are not considered when code retrieval is carriedout, but added separately after the retrieval. To be specific, as shownin FIG. 31(a), the input sub-vector is pattern-compared with the codespossessed by the transmission-side code book 29003, without regard forthe signs (negative or positive) of the 2-bit elements on thelow-frequency side of each sub-vector. For example, there are stored 256codes together with the low-frequency side 2-bit elements, both beingpositive, and the audio code selection unit 290102 retrieves the inputsub-vector and the 256 codes possessed by the transmission-side codebook 29003. Then, any of the combinations shown in FIG. 31(b), which isextracted by the phase information extraction unit 2900107, is added tothe selected code, as signs of the 2 bits on the low-frequency side ofthe sub-vector, and a code index of 10 bits in total is output.

Thereby, the code index output from the audio coding apparatus remainsas in the conventional apparatus, i.e., 10 bits (1024 pieces), but thecode stored in the transmission-side code book 3 can be 8 bits (256pieces). Assuming that the total of the data quantities of the codeindex and the phase information is equal to the data quantity of thecode index for distance calculation shown in formula (23), when thesynthesis sound decoded in formula (23) is compared with the synthesissound according to the embodiment structure, approximately equalsubjective evaluation results are obtained. $\begin{matrix}{D_{i} = {\sum\limits_{i = 0}^{N}\quad {\sum\limits_{j = 0}^{19}\quad \left\{ {{Cij} - {Xj}} \right\}^{P}}}} & (23)\end{matrix}$

Table 3 shows the relationship between the calculation amount and thememory amount in the case where the embodiment structure and formula(22) are used. It can be seen from Table 3 that the structure of thisembodiment reduces the code book to ¼, and reduces the calculationamount to 256 ways of retrieval processes (whereas 1024 ways ofretrieval processes are needed in the conventional structure) and aprocess of adding two codes to the retrieval result, whereby thecalculation amount and the memory are significantly reduced.

TABLE 3 method formula 3 formula 1 transmission data 9 bits 9 bitsquantity code book (number 512 (9 bits) 64 (6 bits) of codes) data forcode 0 3 codes (3 bits) transmission calculation amount 512-codesretrieval 64-codes retrieval ÷ 3-codes addition

As described above, according to the tenth embodiment of the invention,when selecting an audio code having a minimum distance among theauditive distances between sub-vectors produced by dividing an inputvector and audio codes in the transmission-side code book 29003, aportion corresponding to an element of a sub-vector of a high auditiveimportance is treated in the audio code selection unit 2900102 whileneglecting the positive and negative codes indicating its phaseinformation, and subjected to comparative retrieval with respect to theaudio codes in the transmission-side code book 29003. Then, phaseinformation corresponding to an element portion of the sub-vectorextracted in the phase information extraction unit 2900107 is added tothe result obtained, and the result is output as a code index.Therefore, the calculation amount in the audio code selection unit2900102 and the number of codes required in the code book 29003 arereduced without degrading the sensible sound quality.

Embodiment 11

Hereinafter, an audio signal coding apparatus according to an eleventhembodiment of the present invention will be described with reference tothe drawings. FIG. 32(a) is a diagram showing the structure of an audiosignal coding apparatus according to this eleventh embodiment. In FIG.32, reference numeral 3200103 denotes an auditive psychological weightvector table that stores a table of relative auditive psychologicalamounts at the respective frequencies, with regard to the auditivepsychological characteristic of human beings.

Hereinafter, a description is given of the operation. This eleventhembodiment is different from the tenth embodiment in that the auditivepsychological weight vector table 3200103 is newly added. The auditivepsychological weight vectors are obtained by collecting elements in thesame frequency band corresponding to the respective elements of theinput vector of this embodiment from, for example, an auditivesensitivity table defined as auditive sensitivity characteristic tofrequencies, on the basis of the auditive psychological model of humanbeings, and then transforming these elements to vectors. As shown inFIG. 32(b), this table has a peak about a frequency of 2.5 KHz, and thismeans that the elements at the lowest position of frequency are notalways important for the auditive sense of human beings.

To be specific, in this eleventh embodiment, using MDCT coefficients asinput vectors to the audio code selection unit 2900102, and the auditivepsychological weight vector table 3200103 as weights for code selection,auditive distances between the input vectors and the respective codes inthe transmission-side code book 29003 are calculated, and a code indexof a code having a minimum distance is output. When the code index is i,the distance scale Di for code selection in the audio code selectionunit 2900102 becomes, for example, $\begin{matrix}{D_{i} = {{\sum\limits_{i = 0}^{N}\quad {\sum\limits_{j = 0}^{M}\quad {{Wj}\left\{ {{{abs}\quad ({Cij})} - {{abs}\quad ({Xj})}} \right\}^{P}}}} + {\sum\limits_{i = 0}^{N}\quad {\sum\limits_{j = {M + 1}}^{19}\quad {{Wj}\left\{ {{Cij} - {Xj}} \right\}^{P}}}}}} & (24)\end{matrix}$

where N is the number of all codes in the transmission-side code book29003, and Cij is the value of the j-th element in the code index i. Inthis embodiment, M is a number smaller than 19, for example, 1. P is thenorm in the distance calculation, for example, 2. Wj is the j-th elementof the auditive psychological weight vector table 3200103. Further, abs() means absolute operation.

The phase information extraction unit 2900107 decides that phaseinformation of an element corresponding to an audio feature vector ofwhich frequency is extracted the auditive psychological weight vectortable 3200103, and outputs a code index I having a minimum Di in therange and M pieces of phase information Ph(j) j=0 to M.

As described above, according to the eleventh embodiment, when selectingan audio code having a minimum distance among the auditive distancesbetween sub-vectors produced by dividing an input vector and audio codesin the transmission-side code book 29003, a portion corresponding to anelement of a sub-vector of a high auditive importance is treated in theaudio code selection unit 2900102 while neglecting the positive andnegative codes indicating their phase information, and subjected tocomparative retrieval with respect to the audio codes in thetransmission-side code book 29003. Then, phase information correspondingto an element portion of the sub-vector extracted in the phaseinformation extraction unit 2900107 is added to the result obtained, andthe result is output as a code index. Therefore, the calculation amountin the audio code selection unit 2900102 and the number of codesrequired in the code book 29003 are reduced without degrading thesensible sound quality.

Further, the audio feature vector, which is treated in the audio codeselection unit 2900102 while neglecting the positive and negative codesindicating its phase information, is selected after being weighted usingthe auditive psychological weight vector table 3200103 that stores atable of relative auditive psychological amounts at the respectivefrequencies in view of the auditive psychological characteristic ofhuman beings. Thereby, as compared with the tenth embodiment in which aprescribed number of vectors are simply selected from a low band,quantization with more sensible sound quality is realized.

Embodiment 12

Hereinafter, an audio signal coding apparatus according, to a twelfthembodiment of the present invention will be described with reference tothe drawings. FIG. 33(a) is a diagram illustrating the structure of anaudio signal quantization apparatus according to this twelfthembodiment. In the figure, reference numeral 3300104 denotes a smoothingvector table in which data, such as a division curve, are storedactually. Reference numeral 3300105 denotes a smoothing unit thatsmoothes an input vector by division of corresponding vector elements,using the smoothing vector stored in the smoothing vector table 3300104.

Hereinafter, a description is given of the operation. To the smoothingunit 3300105, MDCT coefficients or the like are input as an inputvector, as in the audio signal coding apparatus according to the tenthor eleventh embodiment. The smoothing unit 3300105 subjects the inputvector to smoothing operation using a division curve which is asmoothing vector stored in the smoothing vector table 3300104. Thissmoothing operation is expressed by formula (25) when the input vectoris X, the smoothing vector 3300104 is F, the output from the smoothingunit 3300105 is Y, and the I-th element of each vector is Xi,Fi,Yi.

Yi=Xi/Fi  (25)

When the input vector is MDCT coefficients, the smoothing vector table3300104 is a value that reduces the dispersion of the MDCT coefficients.FIG. 33(b) schematically shows the above-described smoothing process,and the range of data quantity per frequency can be reduced byperforming division of two elements from the low-band side, among theelements transformed to a sub-vector.

The output from the smoothing unit 3300105 is input to the audio codeselection unit 2900102. In the phase information extraction unit2900107, from the smoothed input vector, phase information of twoelements from the lower-frequency side is extracted. On the other hand,in the audio code selection unit 2900102, the smoothed input vector andthe 256 codes stored in the transmission-side code book 330031 areretrieved. Since a correct retrieval result is not obtained if a codeindex (8 bits) corresponding to the obtained retrieval result is outputas it is, information relating to the smoothing process is obtained fromthe smoothing vector table 3300104, and the scaling is adjusted.Thereafter, a code index (8 bits) corresponding to the retrieval resultis selected, and phase information of 2 bits is added to the obtainedresult, thereby to output a coded index I of 10 bits.

The distance Di between the input vector and the code stored in thetransmission-side code book 330031 is expressed by, for example, formula(26) with each i-th element in the smoothing vector table 3300104 beingFi. $\begin{matrix}{D_{i} = {{\sum\limits_{i = 0}^{N}\quad {\sum\limits_{j = 0}^{M}\quad {{Fj}\left\{ {{{abs}\quad ({Cij})} - {{abs}\quad ({Xj})}} \right\}^{P}}}} + {\sum\limits_{i = 0}^{N}\quad {\sum\limits_{j = {M + 1}}^{19}\quad {{Fj}\left\{ {{Cij} - {Xj}} \right\}^{P}}}}}} & (26)\end{matrix}$

where N is the number of all codes in the transmission-side code book330131, and Cij is the value of the j-th element in the code index i. Inthis embodiment, M is a number smaller than 19, for example, 1. P is thenorm in the distance calculation, for example, 2. Wj is the j-th elementof the auditive psychological weight vector table 3200103. Further, abs() means absolute operation. The phase information extraction unit2900107 outputs a code index i having a minimum Di, and M pieces ofphase information Ph(j) j=0 to M. The phase information Ph(j) is definedsimilarly in formula (22).

As described above, according to the twelfth embodiment, when selectingan audio code having a minimum distance among the auditive distancesbetween sub-vectors produced by dividing an input vector and audio codesin the transmission-side code book 330031, a portion corresponding to anelement of a sub-vector of a high auditive importance is treated in theaudio code selection unit 2900102 while neglecting the positive andnegative codes indicating their phase information, and subjected tocomparative retrieval with respect to the audio codes in thetransmission-side code book 330031. Then, phase informationcorresponding to an element portion of the sub-vector extracted in thephase information extraction unit 2900107 is added to the resultobtained, and the result is output as a code index. Therefore, thecalculation amount in the audio code selection unit 2900102 and thenumber of codes required in the code book 330031 are reduced withoutdegrading the sensible sound quality.

Further, since the input vector is smoothed using the smoothing table3300104 and the smoothing unit 3300105, the quantity of data perfrequency, which data are stored in the transmission-side code book330031 to be referred to when the audio code selection unit 2900102performs is retrieval, is reduced as a whole.

Embodiment 13

Hereinafter, an audio signal coding apparatus according to a thirteenthembodiment of the present invention will be described with reference tothe drawings. FIG. 34 is a diagram illustrating the structure of anaudio signal coding apparatus according to this thirteenth embodiment.In the figure, this thirteenth embodiment is different from theembodiment 12 shown in FIG. 33 in that, when the audio code selectionunit 2900102 performs code selection, in addition to the smoothingvector table 3300104, the auditive psychological weight vector table3200103 used for the eleventh embodiment is used as well.

Hereinafter, a description is given of the operation. As in the tenthembodiment, MDCT coefficients or the like are input, as an input vector,to the smoothing unit 3300105, and the output from the smoothing unit3300105 is input to the audio code-selection unit 2900102. In the audiocode selection unit 2900102, the distances between the respective codesin the transmission-side code book 330031 and the output from thesmoothing unit 3300105 are calculated, on the basis of the informationabout the smoothing process output from the smoothing vector table3300104, while adding the weighting by the auditive psychological weightvector in the auditive psychological weight vector table 3200103 andconsidering the scaling in the smoothing process. Using an expressionsimilar to those of the tenth and eleventh embodiments, the distance Diis expressed as, for example, formula (27). $\begin{matrix}{D_{i} = {{\sum\limits_{i = 0}^{N}\quad {\sum\limits_{j = 0}^{M}\quad {{WjFj}\left\{ {{{abs}\quad ({Cij})} - {{abs}\quad ({Xj})}} \right\}^{P}}}} + {\sum\limits_{i = 0}^{N}\quad {\sum\limits_{j = {M + 1}}^{19}\quad {{WjFj}\left\{ {{Cij} - {Xj}} \right\}^{P}}}}}} & (27)\end{matrix}$

where N is the number of all codes in the transmission-side code book330131, and Cij is the value of the lath element in the code index i. Inthis embodiment, M is a number smaller than 19, for example, 1. P is thenorm in the distance calculation, for example, 2. Wj is the j-th elementof the auditive psychological weight vector table 3200103. Further, abs() means absolute operation. The phase information extraction unit2900107 outputs a code index I having a minimum Di, and M pieces ofphase information Ph(j) j=0 to M. The phase information Ph(j) is definedsimilarly in formula (22).

As described above, according to the thirteenth embodiment, whenselecting an audio code having a minimum distance among the auditivedistances between sub-vectors produced by dividing an input vector andaudio codes in the transmission-side code book 330031, a portioncorresponding to an element of a sub-vector of a high auditiveimportance is treated in the audio code selection unit 2900102 whileneglecting the positive and negative codes indicating their phaseinformation, and subjected to comparative retrieval with respect to theaudio codes in the transmission-side code book 330031. Then, phaseinformation corresponding to an element portion of the sub-vectorextracted in the phase information extraction unit 2900107 is added tothe result obtained, and the result is output as a code index.Therefore, the calculation amount in the audio code selection unit2900102 and the number of codes required in the code book 330031 arereduced without degrading the sensible sound quality.

Further, the audio feature vector, which is treated in the audio codeselection unit 2900102 while neglecting the positive and negative codesindicating its phase information, is selected after being weighted usingthe auditive psychological weight vector table 3200103 that stores atable of relative auditive psychological amounts at the respectivefrequencies in view of the auditive psychological characteristic ofhuman beings. Thereby, as compared with the tenth embodiment in which aprescribed number of vectors are simply selected from a low band,quantization with more sensible sound quality is realized.

Further, since the input vector is smoothed using the smoothing table3300104 and the smoothing unit 3300105, the quantity of data perfrequency, which data are stored in the transmission-side code book330031 to be referred to when the audio code selection unit 2900102performs retrieval, is reduced as a whole.

Embodiment 14

Hereinafter, an audio signal coding apparatus according to a fourteenthaspect of the present invention will be described with reference to thedrawings. FIG. 35 is a diagram illustrating the structure of an audiosignal coding apparatus according to this fourteenth embodiment. In thefigure, reference numeral 3500106 denotes a sorting unit which receivesthe output from the auditive psychological weight vector table 3200103and the output from the smoothing vector, selects a plurality of largestelements among the calculated vectors, and outputs these elements.

Hereinafter, a description is given of the operation. This fourteenthembodiment is different from the thirteenth embodiment in that thesorting unit 3500106 is added, and in the method of selecting andoutputting a code index by the audio code selection unit 2900102.

To be specific, the sorting unit 3500106 receives the outputs from theauditive psychological weight vector table 3200103 and the smoothingvector table 3300104 and, when the j-th element of a vector WF isdefined as WFj, it is expressed by formula (28).

WFj=abs(Wj*Fj)  (28)

The sorting unit 3500106 calculates R pieces of largest elements fromthe respective elements WFj of the vector WF, and outputs the numbers ofthe R pieces of element. The audio code selection unit 2900102calculates the distance Di, as in the aforementioned embodiments. Thedistance Di is expressed by, for example, formula (29). $\begin{matrix}{{D_{i} = {\sum\limits_{i = 0}^{N}\quad {\sum\limits_{j = 0}^{19}\quad {FUNCW}}}}{{FUNCW} = \left\{ \begin{matrix}{{Wj}*{Fj}*\left\{ {{{abs}\quad ({Cij})} - {{abs}\quad ({Xj})}} \right\}^{P}} & {at} & {{Rj} = 1} \\{{Wj}*{Fj}*\left\{ {{Cij} - {Xj}} \right\}^{P}} & {at} & {{Rj} = 0}\end{matrix} \right.}} & (29)\end{matrix}$

where, when Rj is the element number output from the sorting unit3500106, Rj is equal to 1 and, when Rj is not the output element number,Rj is equal to 0. N is the number of all codes in the transmission-sidecode book 330031, and Cij is the value of the j-th element in the codeindex i. In this embodiment, M is a number smaller than 19, forexample, 1. P is the norm in the distance calculation, for example, 2.Wj is the j-th element of the auditive psychological weight vector table3200103. Further, abs( ) means absolute operation. The phase informationextraction unit 2900107 outputs a code index I having a minimum Di, andM pieces of phase information Ph(j)j=0 to M. The phase information Ph(j)is defined in formula (30). $\begin{matrix}{{{Ph}(j)} = \left\{ \begin{matrix}{{1\quad {at}\quad {Cji}*{Xj}} \geq 0} \\{{{- 1}\quad {at}\quad {Cji}*{Xj}} < 0}\end{matrix} \right.} & (30)\end{matrix}$

However, Ph(j) is calculated for only those corresponding to the elementnumbers output from the sorting unit 3500106. In this embodiment, (R+1)pieces are calculated. In the case of employing the structure of thisfourteenth embodiment, it is necessary to provide the sorting unit3500106 when decoding this index.

As described above, according to the fourteenth embodiment, in thethirteenth embodiment described above, the output from the smoothingvector table 3300104 and the output from the auditive psychologicalweight vector table 3200103 are received and, from these output results,a plurality of largest elements among the vectors, i.e., elements havinglarge weight absolute values, are selected to be output to the audiocode selection unit 2900102. Therefore, a code index can be calculatedwhile considering both of the elements being significant for theauditive characteristic of human beings and the physically importantelements, whereby coding of a higher-quality audio signal is realized.

While in this fourteenth embodiment R pieces of elements are selectedfrom elements having large weight absolute values with regard to both ofthe smoothing vector 3300104 and the auditive psychological weightvector 3200103, this number may be equal to M used for the tenth tothirteenth embodiments.

Embodiment 15

Hereinafter, an audio signal decoding apparatus according to a fifteenthembodiment of the present invention will be described with reference tothe drawings. FIG. 36 is a diagram illustrating the structure of anaudio signal decoding apparatus according to the fifteenth embodiment.In FIG. 36, reference numeral 360021 denotes a decoding apparatus whichcomprises a receiving-side code book 360061, and a code decoding unit360051. The code decoding unit 360051 comprises an audio code selectionunit 2900102 and a phase information extraction unit 2900107.

Hereinafter, a description is given of the operation. In this fifteenthembodiment, when decoding a code index received, the coding methodaccording to any of the tenth to fourteenth embodiments is applied. Tobe specific, in the audio code selection unit 2900102, for example,elements corresponding to 2 bits from the low-band side, which areauditively important for human beings, are excluded from the 10-bit codeindex received, and the remaining elements corresponding to 8 bits aresubjected to comparative retrieval with the codes stored in thereceiving-side code book 360061. With respect to the excluded 2-bitelements, the phase information thereof is extracted using the phaseinformation extraction unit 2900107, and added to the retrieval result,whereby an audio feature vector is reproduced, i.e., inverselyquantized.

Thereby, the receiving-side code book stores only 256 pieces of codescorresponding to the 8-bit elements, whereby the data quantity stored inthe receiving-side code book 360061 can be reduced. In addition, theoperation in the audio code selection unit 2900102 is 256 times of coderetrieval, and addition of 2 codes to each retrieval result, whereby theoperation amount is significantly reduced.

While in this fifteenth embodiment the structure according to the tenthembodiment is applied to the receiving-side structure, any of thestructures according to the second to fifth embodiments can be applied.Further, when it is used, not independently on the receiving side, butcombined with any of the tenth to fourteenth embodiments, it is possibleto construct an audio data transmitting/receiving system that cansmoothly perform compression and expansion of an audio signal.

Applicability in Industry

As described above, according to an audio signal coding method of thepresent invention, this method is for coding a data quantity by vectorquantization using a multiple-stage quantization method comprising afirst-stage vector quantization process for vector-quantizing afrequency characteristic signal sequence which is obtained by frequencytransformation of an input audio signal, and second-and-onward-stages ofvector quantization processes for vector-quantizing a quantization errorcomponent in the previous-stage vector quantization process: wherein,among the multiple stages of quantization processes according to themultiple-stage quantization method, at least one vector quantizationprocess performs vector quantization using, as weighting coefficientsfor quantization, weighting coefficients on frequency, calculated on thebasis of the spectrum of the input audio signal and the auditivesensitivity characteristic showing the auditive nature of human beings.Therefore, efficient quantization can be carried out by utilizing theauditive nature of human beings.

Furthermore, according to another audio signal coding method of thepresent invention, this method is for coding a data quantity by vectorquantization using a multiple-stage quantization method comprising afirst vector quantization process for vector-quantizing a frequencycharacteristic signal sequence which is obtained by frequencytransformation of an input audio signal, and a second vectorquantization process for vector-quantizing a quantization errorcomponent in the first vector quantization process. In this method, onthe basis of the spectrum of the input audio signal and the auditivesensitivity characteristic showing the auditive nature of human beings,a frequency block having a high importance for quantization is selectedfrom frequency blocks of the quantization error component in the firstvector quantization process and, in the second vector quantizationprocess, the quantization error component of the first quantizationprocess is quantized with respect to the selected frequency block.Therefore, efficient quantization can be carried out by utilizing theauditive nature of human beings.

Furthermore, according to another audio signal coding method of thepresent invention, this method is for coding a data quantity by vectorquantization using a multiple-stage quantization method comprising afirst-stage vector quantization process for vector-quantizing afrequency characteristic signal sequence which is obtained by frequencytransformation of an input audio signal, and second-and-onward-stages ofvector quantization processes for vector-quantizing a quantization errorcomponent in the previous-stage vector quantization process. In thismethod, among the multiple stages of quantization processes according tothe multiple-stage quantization method, at least one vector quantizationprocess performs vector quantization using, as weighting coefficientsfor quantization, weighting coefficients on frequency, calculated on thebasis of the spectrum of the input audio signal and the auditivesensitivity characteristic showing the auditive nature of human beings,and, on the basis of the spectrum of the input audio signal and theauditive sensitivity characteristic showing the auditive nature of humanbeings, a frequency block having a high importance for quantization isselected from frequency blocks of the quantization error component inthe first-stage vector quantization process and, in the second-stagevector quantization process, the quantization error component of thefirst-stage quantization process is quantized with respect to theselected frequency block. Therefore, efficient quantization can becarried out by utilizing the auditive nature of human beings.

Furthermore, according to another audio signal coding apparatus of thepresent invention, this apparatus comprises: a time-to-frequencytransformation unit for transforming an input audio signal to afrequency-domain signal; a spectrum envelope calculation unit forcalculating a spectrum envelope of the input audio signal; anormalization unit for normalizing the frequency-domain signal obtainedin the time-to-frequency transformation unit, with the spectrum envelopeobtained in the spectrum envelope calculation unit, thereby to obtain aresidual signal; an auditive weighting calculation unit for calculatingweighting coefficients on frequency, on the basis of the spectrum of theinput audio signal and the auditive sensitivity characteristic showingthe auditive nature of human beings; and a multiple-stage quantizationunit having multiple stages of vector quantization units connected incolumns, to which the normalized residual signal is input, at least oneof the vector quantization units performing quantization using weightingcoefficients obtained in the weighting unit. Therefore, efficientquantization can be carried out by utilizing the auditive nature ofhuman beings.

Furthermore, according to another audio signal coding apparatus of thepresent invention, in the invention described above, plural quantizationunits among the multiple stages of the multiple-stage quantization unitperform quantization using the weighting coefficients obtained in theweighting unit, and the auditive weighting calculation unit calculatesindividual weighting coefficients to be used by the multiple stages ofquantization units, respectively. Therefore, efficient quantization canbe carried out by effectively utilizing the auditive nature of humanbeings.

Furthermore, according to another aspect of the present invention, themultiple-stage quantization unit comprises: a first-stage quantizationunit for quantizing the residual signal normalized by the normalizationunit, using the spectrum envelope obtained in the spectrum envelopecalculation unit as weighting coefficients in the respective frequencydomains; a second-stage quantization unit for quantizing a quantizationerror signal from the first-stage quantization unit, using weightingcoefficients calculated on the basis of the correlation between thespectrum envelope and the quantization error signal of the first-stagequantization unit, as weighting coefficients in the respective frequencydomains; and a third-stage quantization unit for quantizing aquantization error signal from the second-stage, quantization unitusing, as, weighting coefficients in the respective frequency domains,weighting coefficients which are obtained by adjusting the weightingcoefficients calculated by the auditive weighting calculating unitaccording to the input signal transformed to the frequency-domain signalby the time-to-frequency transformation unit and the auditivecharacteristic, on the basis of the spectrum envelope, the quantizationerror signal of the second-stage quantization unit, and the residualsignal normalized by the normalization unit. Therefore, efficientquantization can be carried out by effectively utilizing the auditivenature of human beings.

Furthermore, according to another audio signal coding apparatus of thepresent invention, this apparatus comprises: a time-to-frequencytransformation unit for transforming an input audio signal to afrequency-domain signal; a spectrum envelope calculation unit forcalculating a spectrum envelope of the input audio signal; anormalization unit for normalizing the frequency-domain signal obtainedin the time-to-frequency transformation unit, with the spectrum envelopeobtained in the spectrum envelope calculation unit, thereby to obtain aresidual signal; a first vector quantizer for quantizing the residualsignal normalized by the normalization unit, an auditive selection meansfor selecting a frequency block having a high importance forquantization among frequency blocks of the quantization error componentof the first vector quantizer; on the basis of the spectrum of the inputaudio signal and the auditive sensitivity characteristic showing theauditive nature of human beings; and a second quantizer for quantizingthe quantization error component of the first vector quantizer withrespect to the frequency block selected by the auditive selection means.Therefore, efficient quantization can be carried out by effectivelyutilizing the auditive nature of human beings.

Furthermore, according to another aspect of the present invention, theauditive selection means selects a frequency block using, as a scale ofimportance to be quantized, a value obtained by multiplying thequantization error component of the first vector quantizer, the spectrumenvelope signal obtained in the spectrum envelope calculation unit, andan inverse characteristic of the minimum audible limit characteristic.Therefore, efficient quantization can be carried out by effectivelyutilizing the auditive nature of human beings. In addition, a portionwhich has been satisfactorily quantized in the first vector quantizer isprevented from being quantized again to generate an error inversely,whereby quantization maintaining a high quality is carried out.

Furthermore, according to another aspect of the present invention, theauditive selection means selects a frequency block using, as a scale ofimportance to be quantized, a value obtained by multiplying the spectrumenvelope signal obtained in the spectrum envelope calculation unit andan inverse characteristic of the minimum audible limit characteristic.Therefore, efficient quantization can be carried out by effectivelyutilizing the auditive nature of human beings. In addition, since thecodes required for quantization can be decreased, the compression ratiois increased.

Furthermore, according to another aspect of the present invention, theauditive selection means selects a frequency block using, as a scale ofimportance to be quantized, a value obtained by multiplying thequantization error component of the first vector quantizer, the spectrumenvelope signal obtained in the spectrum envelope calculation unit, andan inverse characteristic of a characteristic obtained by adding theminimum audible limit characteristic and a masking characteristiccalculated from the input signal. Therefore, efficient quantization canbe carried out by effectively utilizing the auditive nature of humanbeings. In addition, a portion which has been satisfactorily quantizedin the first vector quantizer is prevented from being quantized again togenerate an error inversely, whereby quantization maintaining a highquality is carried out.

Furthermore, according to another aspect of the present invention, theauditive selection means selects a frequency block using, as a scale ofimportance to be quantized, a value obtained by multiplying thequantization error component of the first vector quantizer, the spectrumenvelope signal obtained in the spectrum envelope calculation unit, andan inverse characteristic of a characteristic obtained by adding theminimum audible limit characteristic and a masking characteristic thatis calculated from the input signal and corrected according to theresidual signal normalized by the normalization unit, the spectrumenvelope signal obtained in the spectrum envelope calculation unit, andthe quantization error signal of the first-stage quantization unit.Therefore, efficient quantization can be carried out by effectivelyutilizing the auditive nature of human beings. In addition, a portionwhich has been satisfactorily quantized in the first vector quantizer isprevented from being quantized again to generate an error inversely,whereby quantization maintaining a high quality is carried out.

Furthermore, according to audio signal coding and decoding apparatusesof the present invention, provided for quantization is a structurecapable of performing quantization even at a high data compression ratioby using, for example, a vector quantization method, and employed forallocation of data quantity during quantization is a structure in whichdata contributing to expansion of a reproduced band and datacontributing to improvement of quality are alternately allocated. Firstof all, in the coding apparatus, as the first stage, an input audiosignal is transformed to a signal in the frequency domain, and a portionof the frequency signal is coded; in the second stage, a portion of thefrequency signal uncoded and a coding error signal in the first stageare coded and added to the codes obtained in the first stage; in thethird stage, the other portion of the frequency signal uncoded, andcoding error signals in the first and second stages are coded and addedto the codes obtained in the first and second stages; followed bysimilar coding in forward stages. On the other hand, in the decodingapparatus, both of decoding using only the codes coded in the firststage and decoding using the codes decoded in the first and secondstages are carried out by using the codes decoded in at least the firststage. The decoding order is to decode, alternately, codes contributingto band expansion and codes contributing to quality improvement.Therefore, satisfactory sound quality is obtained even though coding anddecoding are carried out without a fixed data quantity. Further, ahigh-quality sound is obtained at a high compression ratio.

Furthermore, according to another audio signal coding apparatus of thepresent invention, the apparatus comprises: a phase informationextraction unit for receiving, as an input signal, a frequencycharacteristic signal sequence obtained by frequency transformation ofan input audio signal, and extracting phase information of a portion ofthe frequency characteristic signal sequence corresponding to aprescribed frequency band; a code book for containing a plurality ofaudio codes being representative values of the frequency characteristicsignal sequence, wherein an element portion of each audio codecorresponding to the extracted phase information is shown by an absolutevalue; and an audio code selection unit for calculating the auditivedistances between the frequency characteristic signal sequence and therespective audio codes in the code book, selecting an audio code havinga minimum distance, adding phase information to the audio code havingthe minimum distance using the output from the phase informationextraction unit as auxiliary information, and outputting a code indexcorresponding to the audio code having tile minimum distance as anoutput signal. Therefore, the calculation amount in the audio codeselection unit can be reduced without degrading the sensible soundquality. Further, the number of codes to be stored in the code book canbe reduced.

Furthermore, according to another aspect of the present invention, thereis further provided an auditive psychological weight vector table whichis a table of auditive psychological quantities relative to therespective frequencies in view of the auditive psychologicalcharacteristic of human beings, and the phase information extractionunit extracts phase information of an element which matches with avector stored in the auditive psychological weight vector table, fromthe input frequency characteristic signal sequence. Therefore,quantization with improved sensible sound quality is realized.

Furthermore, according to another aspect of the present invention, thereis further provided a smoothing unit for smoothing the frequencycharacteristic signal sequence using a smoothing vector by divisionbetween vector elements and, before selecting the audio code having theminimum distance and adding the phase information to the selected audiocode, the audio code selecting unit converts the selected audio code toan audio code which has not been subjected to smoothing using smoothinginformation output from the smoothing unit, and outputs a code indexcorresponding to the audio code as an output signal. Therefore, thequantity of data per frequency, which data are stored in the code bookand referred to when the audio code selection unit performs retrieval,can be reduced as a whole.

Furthermore, according to another aspect of the present invention, thereare further provided an auditive psychological weight vector table whichis a table of auditive psychological quantities relative to therespective frequencies, in view of the auditive psychologicalcharacteristic of human beings; a smoothing unit for smoothing thefrequency characteristic signal sequence using a smoothing vector bydivision between vector elements; and a sorting unit for selecting aplurality of values obtained by multiplying the values of the auditivepsychological weight vector table and the values of the smoothing vectortable, in order of auditive; importance, and outputting these valuestoward the audio code selection unit. Therefore, it is possible tocalculate a code index while considering both of an element which isimportant for the auditive characteristic of human beings, and anelement which is physically important, resulting in audio signalcompression of higher quality.

Furthermore, according to another audio signal inverse-quantizationapparatus of the present invention, this apparatus comprises: a phaseinformation extraction unit for receiving, as an input signal, one ofcode indices obtained by quantizing frequency characteristic signalsequences which are feature quantities of an audio signal, andextracting phase information of elements of the input code indexcorresponding to a prescribed frequency band, a code book for containinga plurality of frequency characteristic signal sequences correspondingto the code indices, wherein an element portion corresponding to theextracted phase information is shown by an absolute value; and an audiocode selection unit for calculating the auditive distances between theinput code index and the respective frequency characteristic signalsequences in the code book, selecting a frequency characteristic signalsequence having a minimum distance, adding phase information to thefrequency characteristic signal sequence having the minimum distanceusing the output from the phase information extraction unit as auxiliaryinformation, and outputting the frequency characteristic signal sequencecorresponding to the input code index as an output signal. Therefore,the quantity of data stored in the code book used on the receiving endcan be reduced and, further, the calculation amount on the receiving endcan be reduced significantly.

What is claimed is:
 1. An audio signal coding method for coding data,said method being for use with a frequency characteristic signalsequence resulting from frequency transformation of an input audiosignal, said method comprising: a first vector-quantization process forvector-quantizing the frequency characteristic signal sequence, whereinsaid first vector-quantization process produces a quantization errorcomponent; selecting, from frequency bands of the quantization errorcomponent produced by said first vector-quantization process, afrequency band having a highest importance in quantization based on aspectrum of the input audio signal and one or more human auditivesensitivity characteristics; and a second vector-quantization processfor vector-quantizing the quantization error component produced by saidfirst vector-quantization process with respect to only the selectedfrequency band.
 2. An audio signal coding apparatus comprising: atime-to-frequency transformation unit operable to transform an inputaudio signal to a frequency-domain signal; a spectrum envelopecalculation unit operable to calculate a spectrum envelope of the inputaudio signal; a normalization unit operable to normalize thefrequency-domain signal, obtained by said time-to-frequencytransformation unit, with the spectrum envelope, obtained by saidspectrum envelope calculation unit, to obtain a residual signal; a firstvector quantizer operable to vector-quantize the residual signalnormalized by said normalization unit, wherein said firstvector-quantizer produces a quantization error component; an auditiveselection means for selecting, from frequency bands of the quantizationerror component produced by said first vector quantizer, a frequencyband having a highest importance in quantization based on a spectrum ofthe input audio signal and one or more human auditive sensitivitycharacteristics; and a second vector quantizer operable tovector-quantize the quantization error component produced by said firstvector-quantizer with respect to only the frequency band selected bysaid auditive selection means.
 3. An audio signal coding apparatusaccording to claim 2, wherein said auditive selection means selects thefrequency band according to a quantization-importance scale includingvalues obtained by multiplying the quantization error component producedby said first vector quantizer, the spectrum envelope signal obtained bysaid spectrum envelope calculation unit, and the inverse of the humanminimum audible limit characteristic.
 4. An audio signal codingapparatus according to claim 2, wherein said auditive selection meansselects the frequency band according to a quantization-importance scaleincluding values obtained by multiplying the spectrum envelope signalobtained by said spectrum envelope calculation unit and the inverse ofthe human minimum audible limit characteristic.
 5. An audio signalcoding apparatus according to claim 2, wherein said auditive selectionmeans selects the frequency band according to a quantization-importancescale including values obtained by multiplying the quantization errorcomponent produced by said first vector quantizer, the spectrum envelopesignal obtained by said spectrum envelope calculation unit, and theinverse of the sum of the human minimum audible limit characteristic anda masking characteristic calculated from the input signal.
 6. An audiosignal coding apparatus according to claim 2, wherein said auditiveselection means selects the frequency band according to aquantization-importance scale including values obtained by multiplyingthe quantization error component produced by said first vectorquantizer, the spectrum envelope signal obtained by said spectrumenvelope calculation unit, and the inverse of the sum of the humanminimum audible limit characteristic and a masking characteristiccalculated from the input signal and corrected according to the residualsignal normalized by said normalization unit, the spectrum envelopesignal obtained by said spectrum envelope calculation unit, and thequantization error signal produced by said first quantizer.
 7. An audiosignal coding apparatus according to claim 2, wherein said first vectorquantizer is operable to select, from frequency bands of thefrequency-domain signal, a frequency band having a largeenergy-addition-sum of quantization error, and to quantize the selectedfrequency band.
 8. An audio signal coding apparatus according to claim2, wherein said first vector quantizer is operable to select, fromfrequency bands of the frequency-domain signal, a frequency band havinga large energy-addition-sum of quantization error weighted so that afrequency band having a high importance according to the human auditivesensitivity characteristic has a high value, and to quantize theselected frequency band.
 9. An audio signal coding apparatus accordingto claim 2, wherein said first vector quantizer is operable tovector-quantize, at least once, all of the frequency bands of thefrequency-domain signal.
 10. An audio signal coding apparatus accordingto claim 2, wherein said first vector quantizer is operable to calculatea vector quantization error based on a code book, and said second vectorquantizer is operable to vector-quantize the vector quantization errorcalculated by said first vector quantizer.
 11. An audio signal codingapparatus according to claim 10, wherein said first vector quantizer isoperable to calculate the vector quantization error based on a code bookand based on code vectors, all or some of which are inverted.
 12. Anaudio signal coding apparatus according to claim 10, wherein said firstvector quantizer, in calculating the vector quantization error, isoperable to calculate distances for retrieval of an optimum codeaccording to a weighting based on the residual signal.
 13. An audiosignal coding apparatus according to claim 12, wherein said first vectorquantizer, in calculating the vector quantization error, is operable tocalculate distances for retrieval of an optimum code according to aweighting based on the residual signal and according to the humanauditive sensitivity characteristic, and said first vector quantizer isoperable to extract a code having a minimum distance.
 14. An audiosignal coding apparatus according to claim 2, wherein said normalizationunit includes a frequency outline normalization unit operable to roughlynormalize the outline of the frequency-domain signal.
 15. An audiosignal coding apparatus according to claim 2, wherein said normalizationunit includes a band amplitude normalization unit operable to divide thefrequency-domain signal into a plurality of components of continuousunit bands and to normalize the frequency-domain signal by dividing eachunit band with a single value.
 16. An audio signal coding apparatusaccording to claim 2, wherein said first vector quantizer is operable tovector-quantize, at least once, all of the frequency bands of thefrequency-domain signal, and said first vector quantizer includes aplurality of divided vector quantizers operable to separately quantizethe frequency bands of the frequency-domain signal, respectively.
 17. Anaudio signal coding apparatus according to claim 16, wherein: said firstvector quantizer includes, as said plurality of divided vectorquantizers: a low-band divided vector quantizer operable to quantize alow-band component of the frequency-domain signal and to calculate aquantization error of the low-band component, an intermediate-banddivided vector quantizer operable to quantize an intermediate-band ofthe frequency-domain signal and to calculate a quantization error of theintermediate-band component, and a high-band divided vector quantizeroperable to quantize a high-band component of the frequency-domainsignal and to calculate a quantization error of the high-band component;said second vector quantizer is connected after said first vectorquantizer and is operable to quantize a first predetermined band widthof outputs from the divided vector quantizers of said first vectorquantizer; said apparatus further comprises a third vector quantizer,connected after said second vector quantizer, operable to quantize asecond predetermined band width of an output from said second vectorquantizer.
 18. An audio signal coding apparatus according to claim 17,further comprising: a first quantization band selection unit betweensaid first vector quantizer and said second vector quantizer; and asecond quantization band selection unit between said second vectorquantizer and said third vector quantizer; wherein said firstquantization selection unit is operable to select, and input into saidsecond vector quantizer, the outputs from said first vector quantizerthat are in the first predetermined band width, and said second vectorquantizer is operable to quantize the outputs of said first vectorquantizer in the first predetermined band width with respect to thequantization errors calculated by said divided vector quantizers of saidfirst vector quantizer and to calculate, and to output into said secondquantization band selection unit, a quantization error with respect tothe input of said second vector quantizer; and wherein said secondquantization band selection unit is operable to select, and output intosaid third vector quantizer, a portion of the output from said secondvector quantizer that is in the second predetermined band width, andsaid third vector quantizer is operable to quantize the output from saidsecond quantization band selection unit.
 19. An audio signal decodingapparatus for receiving, as an input, codes output from said audiosignal coding apparatus according to claim 17, and for decoding thecodes to output a signal corresponding to the input audio signal, saiddecoding apparatus comprising: an inverse quantization unit operable toperform inverse quantization based only on codes output from saidlow-band divided vector quantizer of said first vector quantizer.
 20. Anaudio signal decoding apparatus for receiving as an input, codes outputfrom said audio signal coding apparatus according to claim 17, and fordecoding the codes to output a signal corresponding to the input audiosignal, said decoding apparatus comprising: an inverse quantization unitoperable to perform inverse quantization based on codes output from saidlow-band divided vector quantizer of said first vector quantizer andbased on codes output from said second vector quantizer.
 21. An audiosignal decoding apparatus for receiving, as an input, codes output fromsaid audio signal coding apparatus according to claim 17, and fordecoding the codes to output a signal corresponding to the input audiosignal, said decoding apparatus comprising: an inverse quantization unitoperable to perform inverse quantization based on codes output from saidlow-band divided vector quantizer and said intermediate-band dividedvector quantizer of said first vector quantizer and based on codesoutput from said second vector quantizer.
 22. An audio signal decodingapparatus for receiving, as an input, codes output from said audiosignal coding apparatus according to claim 17, and for decoding thecodes to output a signal corresponding to the input audio signal, saiddecoding apparatus comprising: an inverse quantization unit operable toperform inverse quantization based on codes output from said low-banddivided vector quantizer and said intermediate-band divided vectorquantizer of said first vector quantizer and based on codes output fromsaid second vector quantizer and codes output from said third vectorquantizer.
 23. An audio signal decoding apparatus for receiving, as aninput, codes output from said audio signal coding apparatus according toclaim 17, and for decoding the codes to output a signal corresponding tothe input audio signal, said decoding apparatus comprising: an inversequantization unit operable to perform inverse quantization based oncodes output from said low-band divided vector quantizer, saidintermediate-band divided vector quantizer, and said high-band vectorquantizer of said first vector quantizer and based on codes output fromsaid second vector quantizer and codes output from said third vectorquantizer.
 24. An audio signal decoding apparatus for receiving, as aninput, codes output from said audio signal coding apparatus according toclaim 16, and for decoding the codes to output a signal corresponding tothe input audio signal, said decoding apparatus comprising: an inversequantization unit operable to perform inverse quantization based oncodes output from some or all of said vector quantizers of said audiosignal coding apparatus.
 25. An audio signal decoding apparatusaccording to claim 24, wherein said inverse quantization unit isoperable to: perform inverse quantization of quantized codes in aprescribed band by executing, alternately, inverse quantization ofquantized codes in a next stage, and inverse quantization of codes in aband different from the prescribed band; continuously execute inversequantization of quantized codes in the different band when there are noquantized codes in the next stage; and continuously execute inversequantization of quantized codes in the next stage when there are noquantized codes in the different band.
 26. An audio signal codingapparatus according to claim 2, wherein: said first vector quantizer isoperable to vector-quantize, at least once, all of the frequency bandsof the frequency-domain signal; said second vector quantizer includes: alow-band divided vector quantizer operable to quantize a low-bandcomponent of the frequency-domain signal and to calculate a quantizationerror of the low-band component, an intermediate-band divided vectorquantizer operable to quantize an intermediate-band of thefrequency-domain signal and to calculate a quantization error of theintermediate-band component, and a high-band divided vector quantizeroperable to quantize a high-band component of the frequency-domainsignal and to calculate a quantization error of the high-band component;said apparatus further comprises a third vector quantizer, connectedafter said second vector quantizer, operable to quantize a predeterminedband width of outputs from said second vector quantizer.
 27. An audiosignal decoding apparatus for receiving, as an input, codes output fromsaid audio signal coding apparatus according to claim 2, for decodingthe codes to output a signal corresponding to the input audio signal,said decoding apparatus comprising: an inverse quantization unitoperable to perform inverse quantization based on at least a portion ofthe codes output from said audio signal coding apparatus to output thefrequency-domain signal; and an inverse frequency transformation unitoperable to transform the frequency-domain signal output by said inversequantization unit into a signal corresponding to the input audio signal.28. An audio signal decoding apparatus for receiving, as an input, codesoutput from said audio signal coding apparatus according to claim 2, andfor decoding the codes to output a signal corresponding to the inputaudio signal, said decoding apparatus comprising: an inversequantization unit operable to reproduce the frequency-domain signal; aninverse normalization unit operable to reproduce the residual signalbased on the codes output by said audio signal coding apparatus, thefrequency-domain signal output by said inverse quantization unit, and tomultiply the residual signal and the frequency-domain signal; and aninverse frequency transformation unit operable to receive an output ofsaid inverse normalization unit and to transform the frequency-domainsignal into a signal corresponding to the input audio signal.